iD808 / iDUCX Version History
Copyright © Speakerbus Technology Ltd. All Rights Reserved.
This document describes the features introduced by each
release, so far as has been recorded. Information about each
release can be reached quickly by following the links provided.
To follow a link, simply click on it.
Version History
From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.520.5.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.520.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.520.5.0
- No features or enhancements added to this release
Defects Resolved in Version 4.520.5.0
- No defects resolved in this release
Known Defects/Issues in Version 4.520.5.0
- When the global muting mode is changed for a pinned appearance tile in iManager the tile disappears from the tile layout on the AYRE device (SB-8046)
- When adding a new speed dial via the iD808 "Speed Dials->Add" menu, the newly created speed dial does not work (SB-7955)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.520.4.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.520.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.520.4.0
- No features or enhancements added to this release
Defects Resolved in Version 4.520.4.0
- VPW Initiated On Speaker on ARIA/AYRE Has No Microphone (when using non-iCS PBX) (SB-7898)
- Unable to add a non-provisioned voice service to the layout (SB-7904)
- Local talker muting stops working on SbRTP voice services after the Aria/Ayre session is temporarily lost (SB-7915)
- At the start of an ARD call, VAD may be incorrectly shown as "on" until there is audio received (SB-7927)
- Moving a key to a different page may cause an ARIA Click key position conflict with an existing key (SB-7936)
- On AYRE, cannot clear ad-hoc group talk off of handset with clear key (SB-7945)
Known Defects/Issues in Version 4.520.4.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.520.3.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.520.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.520.3.0
- Improvements to the message handling when responding to a group talk start or stop message from the ARIA/AYRE UI (SB-7719)
- Process options in the groupTalk message to activate/deactivate the group talk performance changes in the SC (SB-7824)
- Copy sequence number in keepAlive message when responding to keepAlive message (SB-7821)
- Respond to "cmsReconnection" in the "requestInfo" message with a "cmsReconnection" message (SB-7685)
Defects Resolved in Version 4.520.3.0
- When a group talk on a handset includes voice services that are out of service, a reconnection attempt triggered by a SIP re-registration can resulting in the group talk disappearing off the handset on the AYRE UI (SB-7721)
- First call-leg stuck in the on-hold state after the second call-leg is cancelled when transferring a call using TCP SIP signalling (SB-7781)
- When powering up an iD808 in speaker source handset mode, a DMVS Hoot/MRD on a speaker channel may start up with talk latched on (SB-7805)
Known Defects/Issues in Version 4.520.3.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.520.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.520.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.520.2.0
- No features or enhancements added to this release
Defects Resolved in Version 4.520.2.0
- Missed call may be logged when a call has been transferred (SB-7640)
Known Defects/Issues in Version 4.520.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.520.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.520.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.520.1.0
- Spanish Language Support (SB-6889)
- List voice recording streams port number in recording status (SB-7410)
- Hide password when using CM SFTP commands (SB-7309)
- Store passwords in /etc/shadow on the iD808 using SHA512 (SB-7209, SB-7213)
- Remove deprecated SSH Cryptographic Settings on iDUCX (SB-7377)
- Block ICMP timestamp requests on the iD808 (SB-7406)
- DSCP and TTL settings added to configuration message sent to iD924 (SB-7289)
Defects Resolved in Version 4.520.1.0
- On AYRE, the popup 'WARNING Gooseneck not fitted' may be incorrectly displayed (SB-7436)
- Redialling from the call register for a received call logged after picking up an on-hold call, dials the wrong number (SB-7506)
Known Defects/Issues in Version 4.520.1.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.510.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.510.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.510.2.0
- Global muting enhancement for co-located local users (SB-7022)
- Upgrade OpenSSH to 9.8p1 & OpenSSL to 3.0.14 (SB-7076)
- Debug Console 'show ui call_state x' enhanced to list sLeg1RtpData and sLeg2RtpData details.
Defects Resolved in Version 4.510.2.0
- On ARIA/ARYE, the first call is incorrectly put on-hold when a second call is made while the first call is calling out and auto-hold is enabled (SB-6918)
- One way audio after a leg is removed from a conference (SB-6919)
- Cannot seat user at iDUCX if Compliant Call Forwarding is active for the user (SB-6973)
- Compliant Call Forwarding does not work correctly when forwarding to a number that is also bridged on the forwarding device (SB-7066)
- iD808 unusable following unlock compliant call forwarding via seating assistant steps (SB-6986)
- Call is cleared when taking the call off hold after seizing an intercom appearance on a speaker channel (SB-7042)
- CMSIF can crash when iE801s have an IP address (SB-7023)
- One off iD808 crash (SB-6915)
Known Defects/Issues in Version 4.510.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.510.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.510.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.510.1.0
- Latched Hoot Timeout (SB-6342)
- Handle unlocking Compliant Call Forward from iCMS (SB-6461)
- Add support for audioDeviceStatus action 'gt' (group talk) messages (SB-6764)
- Send double tap delay value in optionsConfig from SC to MG (SB-6803)
Defects Resolved in Version 4.510.1.0
- Group talk issue when "Ignore Handset Status" is enabled (SB-6799)
- Group talk mic not active when the handset is active when "Ignore handset status" is enabled (SB-6804)
- Failure to move ARD on Speaker to handset (SB-6830)
- Double clicking or double tapping (very fast) an unlatched (PTT) MRD on a speaker causes the session controller to return an “unsupported functionality” error (SB-6686)
- Command rejected with unknown message type returned when trying to talk on a speaker when no gooseneck mic fitted (SB-6687)
- Internal session controller error seen when dragging an inactive DMVS ARD from speaker to handset (SB-6703)
- Auto select idle handset not being respected when answering a second call via CTI (SB-6627)
- Answering call on an Ayre speaker channel with 'Speaker Source' set to Handset causes the call to enter a 'locked' state (SB-6745)
- With 'Speaker Source' set to handset, when making a call on a speaker channel and then putting the call on handsfree with the handset muted, the handsfree call has the microphone incorrectly muted (SB-6764)
- 'Multiple calls on gooseneck' displayed when putting a single call on handsfree (SB-6771)
- DspProc error messages when making or clearing down DMVS ARD calls from a speaker channel with speaker source set to handset (SB-6773)
- No warning message when handsfree is blocked (SB-6790)
- "Internal Session Controller" error when making an ARD call on a speaker channel (SB-6779)
- Cannot clear an ARD SbRTP call from handset when speaker source is set to 'Handset X' (SB-6815)
- Gooseneck mic lamp is not active when speaker channels are latched on and should be using it (SB-6827)
- ARD on a speaker channel does not move call to handset when 'Speaker Source' is set to 'Handset X' (SB-6836)
- With Speaker Source set to Default Handset, cannot answer a ringing call on a speaker channel (SB-6841)
- Internal session controller error when moving a call from one handset to another (SB-6844)
- Internal session controller error when making a second call on a speaker channel (SB-6845)
- Message protocol error seen when taking a DMVS ARD off hold for first time (SB-6846)
- The speaker source setting should be forced to gooseneck when hosting an ARIA session (SB-6860)
- Placing an ARD call on hold goes to speaker channel incorrectly when 'Speaker Source' is set to 'Handset X' (SB-6859)
- In Ayre, cannot clear a call off a speaker channel after changing speaker source from the UI (SB-6865)
- Cannot barge into a busy-elsewhere DMVS MRD on speaker channel on AYRE located at a local site (SB-6848)
Known Defects/Issues in Version 4.510.1.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.501.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.501.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.501.2.0
- No features or enhancements added to this release
Defects Resolved in Version 4.501.2.0
- Failure to move ARD on Speaker to handset (SB-6830)
Known Defects/Issues in Version 4.501.2.0
- Double clicking or double tapping (very fast) an unlatched (PTT) MRD on a speaker causes the session controller to return an “unsupported functionality” error (SB-6686)
- Command rejected with unknown message type returned when trying to talk on a speaker when no gooseneck mic fitted (SB-6687)
- Internal session controller error seen when dragging an inactive DMVS ARD from speaker to handset (SB-6703)
- Auto select idle handset not being respected when answering a second call via CTI (SB-6627)
- Answering call on an Ayre speaker channel with 'Speaker Source' set to Handset causes the call to enter a 'locked' state (SB-6745)
- With 'Speaker Source' set to handset, when making a call on a speaker channel and then putting the call on handsfree with the handset muted, the handsfree call has the microphone incorrectly muted (SB-6764)
- 'Multiple calls on gooseneck' displayed when putting a single call on handsfree (SB-6771)
- DspProc error messages when making or clearing down DMVS ARD calls from a speaker channel with speaker source set to handset (SB-6773)
- No warning message when handsfree is blocked (SB-6790)
- "Internal Session Controller" error when making an ARD call on a speaker channel (SB-6779)
- Cannot clear an ARD SbRTP call from handset when speaker source is set to 'Handset X' (SB-6815)
- Gooseneck mic lamp is not active when speaker channels are latched on and should be using it (SB-6827)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.501.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.501.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.501.1.0
- No features or enhancements added to this release
Defects Resolved in Version 4.501.1.0
- Group talk issue when "Ignore Handset Status" is enabled (SB-6799)
- Group talk mic not active when the handset is active when "Ignore handset status" is enabled (SB-6804)
Known Defects/Issues in Version 4.501.1.0
- Double clicking or double tapping (very fast) an unlatched (PTT) MRD on a speaker causes the session controller to return an “unsupported functionality” error (SB-6686)
- Command rejected with unknown message type returned when trying to talk on a speaker when no gooseneck mic fitted (SB-6687)
- Internal session controller error seen when dragging an inactive DMVS ARD from speaker to handset (SB-6703)
- Auto select idle handset not being respected when answering a second call via CTI (SB-6627)
- Answering call on an Ayre speaker channel with 'Speaker Source' set to Handset causes the call to enter a 'locked' state (SB-6745)
- With 'Speaker Source' set to handset, when making a call on a speaker channel and then putting the call on handsfree with the handset muted, the handsfree call has the microphone incorrectly muted (SB-6764)
- 'Multiple calls on gooseneck' displayed when putting a single call on handsfree (SB-6771)
- DspProc error messages when making or clearing down DMVS ARD calls from a speaker channel with speaker source set to handset (SB-6773)
- No warning message when handsfree is blocked (SB-6790)
- "Internal Session Controller" error when making an ARD call on a speaker channel (SB-6779)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.500.3.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.500.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.500.3.0
- Disconnect when no longer receiving keep-alive messages from web page (SB-6625)
- Remove 'Playback' key if no speaker channels are configured (SB-6587)
Defects Resolved in Version 4.500.3.0
- Cannot increase SbRTP inactivity timeout on iDUCX (SB-6562)
- CTI busy-elsewhere message timing issue (SB-5990)
- When logging into ARIA, if there are voice services on any speaker channels that are receiving audio, messages to the DSP may be lost (SB-6571)
- When the master volume on an iD808 is set to zero, and a user logs into ARIA, no sound is heard from the speaker channels in the ARIA session (SB-6568)
Known Defects/Issues in Version 4.500.3.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.500.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.500.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.500.2.0
- No features or enhancements added to this release
Defects Resolved in Version 4.500.2.0
- Audio may be heard on muted speaker channels on ARIA/AYRE (SB-6343)
- Automatic level reduction does not work on ARIA/AYRE (SB-6362)
- When multiple telephone-events are offered in the SDP of a SIP INVITE the iDUCX/iD808 can respond with the wrong payload code (SB-6363)
- Possible crash when changing the automatic level reduction setting on a speaker channel (SB-6361)
- "Not allowed when compliant call forwarding is enabled" message shown when signing to ARIA/AYRE and screen it already locked (SB-6352)
- iTurret sends Announce messages to iCMS when ARIA Click signs in or out when using the iTurret as a host (SB-6365)
- Translations for playback mode warning message are not shown in ARIA/AYRE (SB-6419)
Known Defects/Issues in Version 4.500.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.500.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.500.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.500.1.0
- Support enhanced handset behaviour features when hosting an ARIA/AYRE client (SB-6034)
- Support playback to desk on ARIA/AYRE (SB-6160)
- OpenSSL updated to v3.0.12 on iD808
- OpenSSH updated to v9.6p1 on iD808
- Remove deprecated SHA1 settings for SSH (SB-6293)
Defects Resolved in Version 4.500.1.0
- A missed call can be incorrectly logged when a ringing call is ended as a result of the call being forwarded to another number (SB-6212)
- Answering a call on an iD808 when auto-hold mode is set to "auto clear" requires two presses of the dynamic key instead of one (SB-6243)
- Playback screen may be left displayed on screen when updating firmware (SB-6247)
- Intercom call can be made on wrong audio device when intercom line seized (SB-6208)
- Intercom audio device is not always honoured correctly on iTurret (SB-6216)
- CM defaultfactory does not delete Root or SBEngineer history files (SB-6274)
- ARIA devices do not show the line reference on PW appearances in dual line mode (SB-6332)
Known Defects/Issues in Version 4.500.1.0
- Audio on muted speaker channel on ARIA/AYRE (SB-6343)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.400.3.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.400.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.400.3.0
- No features or enhancements added to this release
Defects Resolved in Version 4.400.3.0
- Audio from speaker channels may be heard in screen lock mode on Ayre (SB-6116)
- No audio on speaker channels on Ayre after exiting Compliant Call Forwarding mode (SB-6117)
Known Defects/Issues in Version 4.400.3.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.400.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.400.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.400.2.0
- No features or enhancements added to this release
Defects Resolved in Version 4.400.2.0
- When speaker source is configured for a handset, activating a group talk might result in some speaker channels not transmitting audio (SB-5969)
- Calls may be forwarded when Ayre is in screen lock mode and call forward is disabled (SB-6049)
- Cancelling a call transfer doesn't clear the line causing the other Turret to continue ringing (SB-6048)
- Speed dial key can't be used when adding a new personal directory entry via Aria user programming (SB-6069)
- Idle soft client connections can become disconnected by the iGS (SB-6079)
Known Defects/Issues in Version 4.400.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.400.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.400.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.400.1.0
- User programming support for Aria and Ayre (SB-2129)
- Support Ayre Lock Screen (SB-4952)
- Portuguese, Brazil Language Support (SB-5640)
- Change "Display Mode" to "Label Mode" on the iD808 UI (SB-5491)
- Aria/Ayre diagnostics menu access privilege supported (SB-5920)
- Unauthorised users warning message added to the iD808 when logging in via SSH (SB-5719)
- When primary and backup iCMS server comms requests fail, interleave attempts to connect to iCMS between the primary and backup servers (SB-5760)
Defects Resolved in Version 4.400.1.0
- audioDeviceStatus messages sent for handsets when no handsets configured (SB-5615)
- Failed transfer is not handled cleanly on Aria/Ayre (SB-5468)
- Unable to add first paginating key from ID808 (SB-5363)
- Shared line doesn't show in call activity page in Aria if one user has permissions but no key on profile (SB-5493)
- iD808 sends unnecessary Profile Update for LAST_DATE_TIME to iCMS (SB-5650)
- When the iD808 is configured with a SFTP Diagnostics Server the ‘Send Logs’ option on the main iD808 is greyed out (SB-5682)
- A call can incorrectly be shown as a conference when a Cisco private call is put on hold and taken off hold (SB-5694)
- Compliant call forward call disconnects after a while when using Aria (SB-5746)
- When the call forwarding configuration is changed via Cisco CUCM, the UI on Aria/Ayre is not updated by the live update (SB-5888)
- Slow menu exit doesn’t work on the iD808 (SB-5882)
- When the far end offers a rtpmap for telephone events that does not use 101, the iD808/iDUCX always answers with 101 (SB-5908)
- Latched Group Talk not clearing fully on AYRE/ARIA when speaker source is handset (SB-5962)
- Speaker source setting not used when logging into Aria/Ayre (SB-5968)
Known Defects/Issues in Version 4.400.1.0
- When speaker source is configured for a handset, activating a group talk might result in some speaker channels not transmitting audio (SB-5969)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.300.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.300.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.300.2.0
- No features or enhancements added to this release
Defects Resolved in Version 4.300.2.0
- Privacy can't be enabled again once disabled when calling from an iCS appearance to a Cisco appearance (SB-5534)
- Enabling and disabling privacy on ARIA when calling from a Cisco appearance causes the call timer to reset to 0 (SB-5536)
- Enabling and disabling privacy quickly causes call to get stuck and buttons can't be pressed (SB-5537)
- It may not be possible to add a fixed key using key finder on the iD808 (SB-5573)
- "Default appearance not configured" error sent to ARIA when no handsets are configured (SB-5543)
- Orphaned speaker channels in Aria / Ayre after assigning an appearance to a non-paging speaker channel (SB-5571)
Known Defects/Issues in Version 4.300.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.300.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.300.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.300.1.0
- No features or enhancements added to this release
Defects Resolved in Version 4.300.1.0
- Privacy with Cisco not working (SB-5520)
- Issues with Privacy Handset Default on AYRE/ARIA (SB-5505)
Known Defects/Issues in Version 4.300.1.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.200.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.200.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.200.2.0
- No features or enhancements added to this release
Defects Resolved in Version 4.200.2.0
- When entering an IPv6 address on an iD808 the last digit of the address cannot be seen if it fills up the entire text entry box (SB-5194)
- Two user strings are not translated to the selected locale (Chinese, Japanese or German) (SB-5201)
Known Defects/Issues in Version 4.200.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.200.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.200.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.200.1.0
- Compliant Call Forwarding (SB-2128)
- Multi-language support for Aria / Ayre UI (SB-4749)
- Allow the Call Forward configuration to be changed via Cisco CUCM (SB-5093)
- Addition of Caller ID address in the sent call progress and call notify messages for use with CTI (SB-4975)
Defects Resolved in Version 4.200.1.0
- Incorrect CDR event reported (SB-4819)
- Transferring log files via SFTP fails when using a FQDN, if the DNS lookup for the FQDN returns multiple addresses (SB-4985)
- Cisco Server Side Ad-hoc conferencing does not work when using IPv6 (SB-5010)
- ARIA/Turrets not showing the DDI when a CUCM shared appearance is busy-elsewhere (SB-5089)
- Telephone appearances on fixed keys on the iD808 are shown as out-of-service after a resync / logon (SB-5136)
- Calls on the iD808 or iDUCX on speaker channels 19 and 20 may have issues (SB-5172)
Known Defects/Issues in Version 4.200.1.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.100.5.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.100.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.100.5.0
- No features or enhancements added to this release
Defects Resolved in Version 4.100.5.0
- Transfer doesn't work when using handset 2 (SB-5108)
Known Defects/Issues in Version 4.100.5.0
- Cisco Server Side Ad-hoc conferencing does not work when using IPv6 (SB-5010)
- Transferring log files via SFTP fails when using a FQDN, if the DNS lookup for the FQDN returns multiple addresses (SB-4985)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.100.4.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.100.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.100.4.0
- Update openSSL (to 1.1.1t) and openSSH (to 9.2p1) (SB-5041)
- Do not open the microphone for speaker channels when receiving a makeCall or answerCall message for a speaker channel on Aria / Ayre (SB-5038)
Defects Resolved in Version 4.100.4.0
- Unable to make a VPW call on a speaker channel from Aria / Ayre (SB-4988)
- Auto-answered VPW on a speaker channel automatically opens the microphone for talking (SB-4997)
- VAD indicator stays active for Hoots on speaker channels when they are removed from a conference while the indicator is active in ARIA (SB-4982)
- When 'kicking' a call from a conference on a handset using Aria, the handset appears as idle on the UI (SB-5012)
- "Invalid Operation" warning thrown after attempting to make an SbRTP ARD call immediately after the previous call for the ARD ended (SB-5035)
Known Defects/Issues in Version 4.100.4.0
- Cisco Server Side Ad-hoc conferencing does not work when using IPv6 (SB-5010)
- Transferring log files via SFTP fails when using a FQDN, if the DNS lookup for the FQDN returns multiple addresses (SB-4985)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.100.3.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.100.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.100.3.0
- No features or enhancements added to this release
Defects Resolved in Version 4.100.3.0
- The "Transfer" softkey does not work when pressed on ARIA / AYRE (SB-4893)
Known Defects/Issues in Version 4.100.3.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.100.2.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.100.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.100.2.0
- Gooseneck Microphone Sensitivity (SB-4640)
Defects Resolved in Version 4.100.2.0
- The iD808 is not showing that a Cisco CUCM Server is offline (SB-4616)
- Busy elsewhere shown on the handset when an SbRTP MRD is barged into and privacy mode is enabled at the same time (SB-4681)
- UI may crash causing a core dump if started when no Cisco PBX has been specified (SB-4703)
- When an iD808 is hosting an Aria session, pressing keys on the iD808 can disrupt the calls in Aria (SB-4714)
- The AYRE/ARIA Touch intercom dashboard is missing the group name (SB-4715)
- Local muting isn't applied when initiating an outbound intercom call via the redial button (SB-4727)
- Speaker channels stay muted when a user exits the intercom splash screen by pressing the back button before dialling any numbers (local muting enabled) (SB-4754)
- ARIA / AYRE, listen only voice services on speaker channels may not work (SB-4771)
- SNMP on iD808 misreporting SIP Status (SB-4789)
- When dragging and dropping a HOOT or MRD from one speaker channel to another, this would not succeed, and a warning would be displayed indicating the call was busy (SB-4790)
Known Defects/Issues in Version 4.100.2.0
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.100.1.0
- SBCommon Version 5.1
- SBID808 Version 2.4
SIP Interface Versions in Version 4.100.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.100.1.0
- Handset privacy default supported (SB-3838, SB-3866)
- Handset record on demand default supported (SB-3895)
- iD808 factory default supported (SB-4021)
- Support added for 802.1x for the iD808
- Block use of unencrypted upgraders for the iD808 (SB-3785)
- The iE901 speaker module LED has been changed to a speaker channel volume mute indicator instead of a master volume mute indicator (SB-3979)
- Add the short label to the channel configuration error messages shown under iCMS status (SB-3996)
- Update openSSL (to 1.1.1q) and openSSH (to 9.0p1) on the iD808 (SB-4292)
- World-writable files removed on the iD808 (SB-4321)
- Add packet loss concealment to the received streams from an iD924
Defects Resolved in Version 4.100.1.0
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When answering or assigning to handset a telephone or DMVS PW, when the handset is in the requesting privacy state, privacy is cancelled (SB-3874)
- When failing to barge-in with a "503 Barge-In in Progress" rejection message, the barge-in attempt might not be repeated (SB-3929)
- When enabling/disabling network trace from the iD808 UI menu, pressing 'Save' does not return the user to the previous menu (SB-4011)
- Fix the iD924 slot transmit enable signalling to the iD924 to only enable the left and right gooseneck echo canceller outputs as needed (SB-3988)
- Dialling an intercom call from the call log in ARIA Click to a user who isn’t seated, continues ringing even after the call is cleared off handsets and speakers (SB-4030)
- Transferring conference to a speaker channel with a PBX appearance already assigned causes issues (SB-4059)
- iD808 barge-in to ARD call contains %2 in IPv6 address in altc line (SB-443)
- Treat a multicast address as an invalid static IP address on the iD808 UI configure network menu (SB-4162)
- The CMSIF tag for an IPv6 paired device doesn't match the tag being used by iCMS(SB-4120)
- CM commands that initiate a file transfer using SFTP do not work with an IPv6 address (SB-4171)
- If the 'CM backup' command fails, it still reports 'Backup completed successfully' (SB-4177)
- The iD808 cancels outgoing calls after 3 minutes (SB-4180)
- iCB Mixer levels incorrect for handsfree operation (SB-4119)
- Taking a private call off hold causes the call to be not private and unable to be made private until it is cleared down (SB-4211)
- Possible MTFIF crash in the iD808 (SB-4243)
- iD808 Linux Security Vulnerabilities (SB-4284)
- When signing into Aria Hoot the Speaker Page is not forced to Speaker Page 1 (SB-4356)
- Paired device doesn't work after synchronizing (SB-4376)
- Improvements to the operation of the dhcpcd client (SB-4427)
- DTMF stops working after unconfigured voicemail button is pressed on Aria (SB-4421)
- MCC messages ignored when using IPv6 (SB-4440)
- Cannot communicate with a Cisco CUCM PBX when using IPv6 (SB-788)
- SIP error when configured for IPv6 with TCP transport (SB-4586)
- Log off from an iD808 causes the device to go out-of-sync in iManager (SB-4579)
- iD808 to iCMS Communications is lost if IPv6 is enabled by a LogOn request message (SB-4592)
- Issue when enabling IPv6 on an iD808 when using Cisco appearances (SB-4619)
- Upgrade / sendlogs commands failing for iDUCX/iCB (using SFTP) (SB-4230)
- iD808 sending out DNS queries when it should not be doing so (SB-4622)
Known Defects/Issues in Version 4.100.1.0
- The iD808 is not showing that a Cisco CUCM Server is offline (SB-4616)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.010.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 4.010.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.010.1.0
- iE901 speaker module LED changed to a speaker channel volume mute indicator instead of a master volume mute indicator (SB-3979)
Defects Resolved in Version 4.010.1.0
- No defects resolved in this release
Known Defects/Issues in Version 4.010.1.0
- Unable to discover devices connected to a switch using LLDP (F0039009)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.000.3.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 4.000.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.000.3.0
- No features or enhancements added to this release
Defects Resolved in Version 4.000.3.0
- The 'Send Logs' option in the 'Main' menu on the iD808 fails if you have previously used the 'Log Upload request' from iManager (SB-3886)
- The iDUCX logs downloaded from an iCB web page do not include the SIP status and some other useful status files (SB-3939)
Known Defects/Issues in Version 4.000.3.0
- Unable to discover devices connected to a switch using LLDP (F0039009)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.000.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 4.000.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.000.2.0
- Allow the use of G.711 and G.722 audio codecs for the iD924 media connection, for iD924 firmware revisions 1.010 and later (SB-3100)
- Support added for interworking with an Avaya PBX using TCP transport for the SIP messages (SB-845)
- iD808 pbx.xml iCMS file configuration updated to include IPv6 server addresses (SB-2747)
- Profile version increased to 25 (indicates support for call duration times in call_register.xml and IPv6 servers in pbx.xml)
- Aria and Ayre call duration support (SB-3520) - Requires iWS v2.6 or later
- Add “cloudBaseServerIpAddress” to the “sessionStatus” message (SB-3672) - Requires iWS v2.6 or later
- Send speaker channel latching configuration to the soft client UI (SB-3750) - Requires iWS v2.6 or later
Defects Resolved in Version 4.000.2.0
- When in Broker Mode the iD808 UI can lockup (SB-3756)
- Several 'DSPIn message queue has blocked' messages being logged for the iD808 (SB-3759)
- No audio heard on speaker channels when using ARIA if the master volume knob is turned down when ARIA connects to the iD808 (SB-3773)
- iD808 losing IPv6 address when user is unseated (SB-3603)
- iD808 sometimes reports its link-local IPv6 address (SB-3604)
- When a DMVS Hoot/MRD is cleared off all audio devices, the internal call data is destroyed (SB-3629)
- Memory tracking and memory leak issues (SB-3601)
- Memory leak seen during running of lots of small autoscripts (SB-3547)
- When the 'Diagnostic Server' configuration for an iD808 is configured as an IPv4 IP address, the 'Send Logs' menu item on the iD808 is greyed out (SB-3666)
- Cannot unlock a ringing ARD voice service (SB-3668)
- iD924 loudspeaker audio levels low (SB-3794)
- CM and iCB help text formatting incorrect (SB-3657)
- Blank line shown at the top of the Network Status screen on the iD808 (SB-3705)
- Error message logged when putting a DMVS ARD on hold (SB-3664)
- With an intercom call on the intercom audio device, dragging the intercom appearance on to a speaker channel leaves the microphone muted, but shown as unmuted on the intercom audio device (SB-3709)
- iCB cannot process FILE_TRANSFER_* CM variables for SFTP support (SB-3730)
- Send logs doesn't delete old temporary files on the iDUCX before creating a new one, increasing the memory usage (SB-3542)
- Conferences in group talk do not look like they are activating when the group talk button is pressed (SB-3765)
- Removing all voice services from a conference on a speaker channel can leave it showing as orphaned on the Aria UI (SB-3762)
- Programming a group talk for an alerting SBRTP ARD speaker channel causes it to go "busy elsewhere" (SB-3864)
Known Defects/Issues in Version 4.000.2.0
- Unable to discover devices connected to a switch using LLDP (F0039009)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 4.000.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 4.000.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 4.000.1.0
- Live update support for Aria and Ayre
- Group talk support for Aria and Ayre
- SFTP supported as an alternative to TFTP for upgrading, transferring logs, etc.
- iCB server name and iCB collection name added to sessionStatus message (SB-2959)
- iCB software version added to the sessionStatus message (SB-3237)
- Allow the use of G.711 and G.722 audio codecs for the iD924 media connection, for iD924 version 1.100 and later (SB-3100)
Defects Resolved in Version 4.000.1.0
- Multiple memory leaks (SB-3299, SB-3407, SB-3408)
- Calls on alias'd lines are incorrectly orphaned when deleting one of the aliases (SB-3284)
- Internal session controller error reported on ARIA / AYRE when attempting to make a VPW call with outbound calls configured as none (SB-3201)
- Possible crash in the SIP stack when receiving truncated packets from the network (SB-3178)
- In Aria Hoot, DMVS ARD calls will not ring if play after X seconds is selected on the alert profile or will not wait the X number of seconds (SB-2886)
- In Aria Hoot, Enabling and disabling DnD stops DMVS calls from ringing (SB-2872)
- Recording Streams are not turned off when a user has logged out / unseated on a iD808 or Aria Click H/W hosted (SB-2818)
- Dragging a VPW appearance tile to a handset tile says "functionality not supported" (SB-2764)
- Security vulnerabilities in various CM commands (SB-2385)
- Multiple errors for broken corporate directory sent in one line to ARIA/AYRE (SB-2797)
- iD808 aes upgrader files are not removed from the iD808 file system if their file name does not include '808_' (SB-3106)
- IPv6 bug fixes
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
- CM sendlogs doesn't delete old temporary file on Turret before creating a new one which can result in the iD808 running out of memory (SB-3524)
- When adding a new paginating key from the iD808 device, the key is not saved in iCMS (SB-3549)
- Adding/Deleting paginating keys via the turret too quickly can cause the iD808 to lock up (SB-3462)
- Cannot clear down an intercom call on a speaker channel in Aria Touch or Ayre after dragging the call from the appearance key to the speaker channel (SB-3727)
Known Defects/Issues in Version 4.000.1.0
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When in Broker Mode the iD808 UI can lockup (SB-3756)
- Several 'DSPIn message queue has blocked' messages being logged for the iD808 (SB-3759)
- No audio heard on speaker channels when using ARIA if the master volume knob is turned down when ARIA connects to the iD808 (SB-3773)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.803.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.803.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.803.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.803.1.0
- When in Broker Mode the iD808 UI can lockup (SB-3756)
- Several 'DSPIn message queue has blocked' messages being logged for the iD808 (SB-3759)
Known Defects/Issues in Version 3.803.1.0
- The Spansion Flash IC (S34ML02G100TFI000) type is reported as unknown by the Kernel
or uBoot (F0026511)
- When a call is dialled from a turret registered to an Avaya PBX to a turret registered on a Cisco v9
PBX, answered and put on hold the CLI displayed on the turret at the Avaya end is not a dial number
but an internal PBX string (F0027366)
- Dialling a speed dial to a number containing '+' changes the label to the number and not the
speed dial label (F0028357)
- DTMF tone levels too low when interfacing to iG214 and analogue phone lines (F0031926)
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- When iCS is unavailable, the iD808 produces error messages reporting the subscription index to be 65535 (F0035533)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
- No audio heard on speaker channels when using ARIA if the master volume knob is turned down when ARIA connects to the iD808 (SB-3773)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.802.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.802.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.802.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.802.1.0
- The DspProc message queue (from the UI) can lock up when handling a link state change message (SB-3318)
Known Defects/Issues in Version 3.802.1.0
- The Spansion Flash IC (S34ML02G100TFI000) type is reported as unknown by the Kernel
or uBoot (F0026511)
- When a call is dialled from a turret registered to an Avaya PBX to a turret registered on a Cisco v9
PBX, answered and put on hold the CLI displayed on the turret at the Avaya end is not a dial number
but an internal PBX string (F0027366)
- Dialling a speed dial to a number containing '+' changes the label to the number and not the
speed dial label (F0028357)
- DTMF tone levels too low when interfacing to iG214 and analogue phone lines (F0031926)
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- When iCS is unavailable, the iD808 produces error messages reporting the subscription index to be 65535 (F0035533)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.801.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.801.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.801.2.0
- Update to openssl-1.0.2k-24 for iDUCX devices
Defects Resolved in Version 3.801.2.0
- No defects resolved in this release
Known Defects/Issues in Version 3.801.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.801.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.801.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.801.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.801.1.0
- Memory leak when processing live updates containing directory changes (SB-3024)
Known Defects/Issues in Version 3.801.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.800.3.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.800.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.800.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.800.3.0
- Seizing an intercom appearance on a speaker causes wrong mappings to be sent to Aria/Ayre (SB-2688)
Known Defects/Issues in Version 3.800.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.800.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.800.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.800.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.800.2.0
- DMVS ARDs do not alert when in remote mode in ARIA Hoot. The call comes though fine however there is no alert sound. This only affects ARIA Hoot (F0039075)
- When you login to ARIA Hoot for the first time, it will show that you have SIP errors even if there is no configuration errors. When you press the SIP status button there will be no errors displayed. If you wait around 20 seconds it will eventually turn green (F0039079)
- Broken voice services on speaker channels 9-24 are being displayed in the SIP Server Errors in ARIA Hoot when they should be ignored (F0039081)
Known Defects/Issues in Version 3.800.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.800.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.800.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.800.1.0
- Support added for personal directory editing from Aria/Ayre
- Support added for Aria Hoot
- Always use TCP for transmitting soft client interface NOTIFY messages
- Send live update action messages to the soft client interface (SB-2518)
- Modify autoscript to allow for names within single quotes - this will allow the use of names that include spaces and other characters
Defects Resolved in Version 3.800.1.0
- White noise heard when voice services are conferenced on the iCS (SB-1630)
- When configuring the call forwarding address in Aria, if a double quote is included in the address string, the change gets accepted by the iD808/iDUCX device, but the response is not sent back to the Aria UI. Further more, if the user then logs out and logs in again, the call forwarding icon will not be displayed on the Aria UI, as the call forward status message still fails to be sent by the device (F0039042)
- On Aria Touch or Ayre, a tile that should be on a virtual page may not appear on the UI (F0039046)
- Answering a VPW on Ayre using the 'answer' key does not show the call on the handset (F0039051)
Known Defects/Issues in Version 3.800.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.700.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.700.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.700.2.0
- Live updates of the personal directory are now supported when in an Aria/Ayre session
Defects Resolved in Version 3.700.2.0
- When going into the key edit menu from key finder, the text on the save/exit button is drawn with the wrong font size (F0039003)
- Able to put an alias and its appearance onto the same page using the turret menu keys (F0039008)
- Once created you cannot delete a VPW alias from a turret (F0039010)
- If you create a personal directory entry in ARIA Click/Touch/ARYE and wait for the live update to be seen in iManager. If you then delete the personal directory entry from iCMS and create a new personal directory entry from ARIA again, it will restore the one you just deleted (F0039016)
- If you import/create a directory whilst connected to ARIA and then create a personal directory entry from ARIA, it will delete every personal directory entry that you created via iCMS (F0039018)
- If you have a key alias and using an ARIA soft client, you change the style of the key, the display of both keys changes, but when opening the key in the editor dialog, the style of the alias is still the same as it was before the change (F0039020)
- If you try to move a key from a fixed position over a paginated key via the turret, the turret will be thrown out of sync (F0039021)
- When changing an alert of a user seated on a turret on a hosted ARIA click session, the live update is not processed correctly. (F0039027)
- You are able to create a personal directory entry with no Short Label and no Address (F0039031)
- When editing an alert via ARIA, you are able to give it a blank name (F0039033)
- You are able to set up call forwarding with no address on ARIA (F0039034)
- Key finder on the iD808 may leave the user on a page or move the user to a page where no keys are highlighted (F0039037)
- In key finder on the iD808, the 'Move' option is not greyed out for 'Speakers' when there is no suitable destination key to move to (F0039040)
- Using a G.711 audio codec for the Ayre connection will result in one-way audio and using a G.722 audio codec for the Ayre connection will crash the iCB server (F0038933). The fix for this is a temporary workaround that forces the 16kHz PCM audio codec to be used, regardless of the iManager setting.
Known Defects/Issues in Version 3.700.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Unable to discover devices connected to a switch using LLDP (F0039009)
- When configuring the call forwarding address in Aria, if a double quote is included in the address string, the change gets accepted by the iD808/iDUCX device, but the response is not sent back to the Aria UI. Further more, if the user then logs out and logs in again, the call forwarding icon will not be displayed on the Aria UI, as the call forward status message still fails to be sent by the device (F0039042)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.700.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.700.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.700.1.0
- Key Alias
- Record-on-demand
- The capabilities reported by the Cisco Discovery Protocol (CDP) for the iD808, changed from "Host" to "VoIP Phone"
Defects Resolved in Version 3.700.1.0
- Memory leak in MTFIF when a barge-in attempt is rejected with a 503 error response (F0038927)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
- If the handset is in handsfree mode, and idle, you can't make intercom calls on Ayre (F0038942)
- iD808 mishandling of some OIDs when using an SNMP-GET, which can cause the iD808 to run out of memory and stop working (F0038943)
- iD924 disconnects when trying to connect a DMVS voice service that is out-of-service using a handset (F0038960)
- Transmit audio muted for intercom call after moving from intercom handsfree to handset (F0038969)
- Incorrect display for Group Calls in Aria Click, Aria Touch and Ayre when the device is a Remote device. You see 'Sites: 0'. Also, "Sites" is not shown on the iD808 (F0038984)
- Adding intercom appearances via key finder may not work correctly when there is no intercom number configured (F0038987)
- Networking issues caused by ARP request failures on the iD808/iE801 (F0038990)
- If you add an intercom appearance via the turret onto a fixed key (make sure that the intercom appearance does not have a dial number), you will be unable to delete that intercom appearance via the turret (F0038991)
Known Defects/Issues in Version 3.700.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.620.6.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.620.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.620.6.0
- No features or enhancements added to this release
Defects Resolved in Version 3.620.6.0
- Using a G.711 audio codec for the Ayre connection will result in one-way audio and using a G.722 audio codec for the Ayre connection will crash the iCB server (F0038933). The fix for this is a temporary workaround that forces the 16kHz PCM audio codec to be used, regardless of the iManager setting.
Known Defects/Issues in Version 3.620.6.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Memory leak in MTFIF when a barge-in attempt is rejected with a 503 error response (F0038927)
- When G.711 or G.722 is selected as the audio codec for the Ayre connection in iManager, the 16kHz PCM codec is used instead of the configured setting (F0038933)
- iD808 mishandling of some OIDs when using an SNMP-GET, which can cause the iD808 to run out of memory and stop working (F0038943)
- iD924 disconnects when trying to connect a DMVS voice service that is out-of-service using a handset (F0038960)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.620.5.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.620.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.620.5.0
- No features or enhancements added to this release
Defects Resolved in Version 3.620.5.0
- iD808/IDUCX crash when multiple calls are using the gooseneck microphone with an Aria / Ayre session active (F0038941)
Known Defects/Issues in Version 3.620.5.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Memory leak in MTFIF when a barge-in attempt is rejected with a 503 error response (F0038927)
- iD808 mishandling of some OIDs when using an SNMP-GET, which can cause the iD808 to run out of memory and stop working (F0038943)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.620.4.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.620.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.620.4.0
- Add IPv6 support to tftp command on the iD808 (SB-1608)
Defects Resolved in Version 3.620.4.0
- When upgrading, old encrypted upgrader files (.tar.gz.aes) are not deleted from the file system (F0038916)
- lldpd not stopped prior to an iD808 upgrade (F0038917)
- CTI does not send the current scId or mgId in the makeCall message for an Idle Hoot or MRD call and as a result the iD808/iDUCX rejects the message if it is not the first makeCall message for the voice service (F0038873)
- The RTP voice recording streams may not be transmitted (F0038920)
- Answering a ringing call on a handsfree handset on Ayre with the handsfree microphone configured as gooseneck result in the handset tile not showing the call (F0038923)
Known Defects/Issues in Version 3.620.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.620.3.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.620.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.620.3.0
- Chinese and Japanese translations updated
- Container status added to sendlogs
Defects Resolved in Version 3.620.3.0
- No audio heard on one leg when conferencing together DMVS voice services (F0038903)
- On Ayre, when barging into a DMVS ARD that`s in a busy elsewhere state and then dropping out of the call, back to the busy-elsewhere state, the caller ID for the call on the call activity tile may show the wrong label (F0038904)
- Some of the line labels on the iD808 are show as name/0 instead of just name (F0038913)
- Bridging a call on handset 1 to handset 2, on the iD808, results in no audio on handset 2 (SB-1545)
- The working iCMS IP address is not cleared when the setting from iCMS is cleared (SB-1551)
Known Defects/Issues in Version 3.620.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.620.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.620.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.620.2.0
- First official support for Ayre (messaging protocol 10)
- Support encrypted upgrades
- Update to openssh-8.5p1
- Update to openssl-1.1.1k
- Support "Prefix" style line labels, in addition to the classic style on the iD808
- Improved M_INFO reporting for iCMS CheckProfile responses
Defects Resolved in Version 3.620.2.0
- Issues setting up the IP address on a turret with a static IP address, from the turret using the keypad (F0038579)
- A potential iD808 crash if an inbound group call is received with no 'from' identity present in the SIP header of the INVITE (F0038769)
- During the iCS failover, endpoint Registrations are failing when the media is over TCP (SB-614)
- Attempting to assign a seized intercom line from a handset to the intercom handsfree device in ARIA does not work (F0038845)
- When dialling a telephone call on the CTI test tool page connected to an iD808, if you have intercom selected, the telephony call will still be made on the turret but the UI shows intercom (F0038850)
- When a turret is logged into the CTI test tool page with call alerts enabled, if you manually put an intercom call on hold (through the turret, not CTI), then take that call off hold (again thorough the turret), the CTI test tool page will take around 30 - 60 seconds to update the call status (F0038851)
- In Aria, when a line is seized and then a number is dialled from the dialpad, the iD808/iDUCX incorrectly applies the outbound dial rules (F0038854)
- The prompt bar shown for an outbound locked group call on the intercom splash screen for an iD808 is incorrect (F00388857)
- Issue with updates to the corporate address book when the iCB has been synchronised more than 255 times since it was last restarted (Zendesk-6469)
- Using CTI, if you attempt to take a telephone call off hold using the answerCall command with the audio device set to intercom, it will allow the command to go through (F0038868)
- Using CTI, if you attempt to initiate a SbRTP voice service using the makeCall option with intercom selected as the audio device, the turret UI will crash. If you try this with DMVS voice services the turret will not crash but it will make the voice service call as an intercom which should be blocked (F0038870)
- If an ARD or an MRD is being alerted to a CTI connected device and you attempt to answer the call, but you select intercom as the audio device, it will allow the command instead of being blocked (F0038872)
- 'If you have an active telephony or voice service call and then change it between headset and speaker from the physical endpoint. There is no update made on the CTI webpage (F0038876)
- Cisco adhoc conferencing fails with using TCP SIP signalling (SB-1483)
- When doing a network trace on an iDUCX, some packets may be truncated
- The Time Zone is not updated via iCMS on iDUCX devices
- IPv6 bug fixes
Known Defects/Issues in Version 3.620.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Bridging a call on handset 1 to handset 2, on the iD808, results in no audio on handset 2 (SB-1545)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.620.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.620.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.620.1.0
- Provisional support for Ayre (messaging protocol 10)
Defects Resolved in Version 3.620.1.0
- No defects resolved in this release
Known Defects/Issues in Version 3.620.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.610.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.610.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.610.2.0
- "Speaker Activity Indication Timeout" configuration supported for iDUCX devices
- Soft Client messaging protocol enhancements to support future features. Should only be utilised after the
protocol version is increased to 10 or higher in a future release.
- Handsfree control supported for Ayre
- Internal microphone and gooseneck indicators implemented for Ayre
- "optionsConfig" message implemented
- "active" added to "keyConfig" and "virtualKeyConfig" messages for appearance keys
- "ro" and "allowAlertOverride" added to the "keyPageConfig" message
Defects Resolved in Version 3.610.2.0
- A barge in fails for a DMVS gateway based ARD call (F0038736)
- "Internal session control error" seen on Aria when adding a conferencing member, if Avaya adhoc conferencing
is enabled and adding an Avaya telephone call to another Avaya telephone call (F0038731)
- Cannot clear CDR IP address strings from iCMS (F0038737)
- IPv6 bug fixes
Known Defects/Issues in Version 3.610.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.610.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.610.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.610.1.0
- LLDP/CDP (Link-layer discovery protocol/Cisco discovery protocol v1 & v2) supported by the iD808.
- Device details advertised to switch using LLDP, CDPv1 or CDPv2.
- Device details reported in SNMP under LLDP-MIB.
- 'ldpcli show interfaces' and 'lldpcli show neighbors' status added to sendlogs.
- iD808 MIB OID reported in SNMP
- Do not send out the caller ID label as 'Conference' to the Aria/CTI interface for a client-side
conference. This gives Aria / CTI the flexibility of using the call state to display 'Conference'
or display the caller ID label of the individual calls that make up the client-side conference.
- For CTI makeCall and answerCall, when the audio device is 'default', if the appearance is an
intercom appearance and the intercom audio device is set to handsfree, the handsfree intercom
device is selected, otherwise the default handset device is selected.
- Change from using SIP port funnelling, for supporting extra UDP and TCP ports when communication to the
Cisco CUCM PBX, to opening the range of ports in the SIP stack.
- Soft Client messaging protocol enhancements to support future features. Should only be utilised after the
protocol version is increased to 10 or higher in a future release.
- Add master volume to audioDeviceAction and audioDeviceStatus for Ayre.
- Support new messages personalDirConfig and alertConfig.
- Handle inbound 'keyConfig' messages.
- Add 'alertEnabled' and 'alert' to 'keyConfig' and 'virtualKeyConfig' messages for appearance keys.
- Add 'address' to the 'appearances' messages.
Defects Resolved in Version 3.610.1.0
- Paired device, CDR server and logs TFTP addresses cannot be cleared to a blank address from iCMS.
- Sequence number not set for Soft Client intercomPrivacyConfig and intercomConfig messages.
- IPv6 bug fixes
Known Defects/Issues in Version 3.610.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.600.3.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.600.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.600.3.0
- Support iGS backwards compatibility for CTI intercom support
- Do not send out the caller ID label as 'Conference' to the Aria/CTI interface
for a client-side conference. This gives Aria / CTI the flexibility of using
the call state to display 'Conference' or the caller ID label to display the caller
ID of the individual calls that make up the client-side conference
Defects Resolved in Version 3.600.3.0
- Internal session controller error seen in ARIA when trying to add a member to a conference (F0038587)
Known Defects/Issues in Version 3.600.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.600.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.600.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.600.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.600.2.0
- When configured for TCP SIP signalling, SIP NOTIFYs sent to the iGS for Aria/CTI signalling and SIP NOTIFY's
sent to iCS always create a new TCP connection instead of re-using any existing connection. This may result
in the MTFIF SIP stack running out of connections and therefore failing to send some SIP NOTIFY messages (F0038558)
Known Defects/Issues in Version 3.600.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.600.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.600.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.600.1.0
- Supports ARIA conferencing
- Supports CTI for intercom and group calls
Defects Resolved in Version 3.600.1.0
- An occassional iDUCX crash when logging out a user (F0038502)
- VLAN tagging does not work on iD808s (F0038524)
- A change to the SNMP configuration may have no affect (F0038537)
Known Defects/Issues in Version 3.600.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.551.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.551.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.551.1.0
- Updated Japanese translations
- iD808 kernel version to 3.550.2.0 to implement RFC5961 (Send a challenge ACK if receive ACK or RST in TCP window, but not with next sequence number). This resolves Qualys QID 82054
Defects Resolved in Version 3.551.1.0
- In Aria, during a successful call transfer via an Avaya PBX, the Aria UI may display "Invalid number" (F0038509)
- An iD808/iDUCX crashed when deleting a Cisco VPW appearance in iCMS and then disassociating the number from the device/user in the Cisco CUCM (F0038511)
- In iD808, MCC not working when enabled after being disabled for many days (e.g. switching from remote to local devices) (F0038513)
Known Defects/Issues in Version 3.551.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.550.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.550.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.550.1.0
- Updated Chinese translations
Defects Resolved in Version 3.550.1.0
- It is possible to get an intercom call stuck on the intercom handsfree device (F0038429)
- When logging into an Aria session, if the iCMS profile errors reported results in a string length that is greater than 8192,
the sessionStatus message sent to the Aria UI will fail to be sent and the HEALTH STATUS screen will be blank (F0038432)
- When setting up an answerback group call and after the caller has pressed Initiate Answerback, moving the call from intercom
to handset, audio is restored when you would expect it to be muted (F0038435)
- When originating an answerback group call on an iD808 from the intercom handsfree device, the call starts with the microphone
active but the handsfree LED is red instead of green (F0038436)
- The “CM help” command for iDUCX devices returns help for “CM logviewer” and “CM logtail” which are not available on an iDUCx (F0038437)
- UI killed by SIGSEGV” when doing an iCB status (F0038447)
- When the CDR server is configured from off to on, a SNMP trap and remote syslog is always triggered showing the CDR connection
as down before changing to connected (F0038449)
- When there is a 3-way conference (us and two far end participants) on a handset with the handset microphone muted, if one of the
far end participants drops out, the handset microphone is active, even though it is shown as muted on the UI (F0038461)
- Turrets may crash if a DNS server IP address is not configured (F0038475)
Known Defects/Issues in Version 3.550.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.540.3.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.540.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.540.3.0
- SNMP sysDescr (MIB II system description) string modified to report the Speakerbus product ID and software version number
Defects Resolved in Version 3.540.3.0
- On Aria, if the handset microphone is muted when the call is cleared, the handset muted state is changed to active on the UI when it is still muted (F0038416)
Known Defects/Issues in Version 3.540.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.540.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.540.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.540.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.540.2.0
- There are issues with Cisco appearances that may stop Cisco calls functioning correctly (F0038404 & F0038412)
- Voice recording may stop working on an iDUCX device (F0038413)
- Ethernet Down and DHCP status error messages are not shown with useful text on the Aria interface. For example "Ethernet NET 2 Down" is just reported as "Down" (F0038414)
Known Defects/Issues in Version 3.540.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.540.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.540.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.540.1.0
- Support added for SNMP V3
- Appearance array increased in size to 900, to accomodate a user with permissions to a higher number of lines
- Subsubscription array increased in size to 2400, to accomodate a user with permissions to a higher number of lines
- Qualys QID 38666, 38739, 78030 & 105459 addressed for iDUCX devices
Defects Resolved in Version 3.540.1.0
- The iD808 does not perform checks to see if Voice Recorder address is resolvable (F0038273)
- Issues when starting iD808 with no network connection (F0038321)
- Issues with ARP not working due to having invalid network interface name (F0038364)
Known Defects/Issues in Version 3.540.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.531.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.532.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.532.1.0
- Japanese translations updated
Defects Resolved in Version 3.532.1.0
- Error messages seen in log file when synchronising if the device is in a
call region with no iCS server (F0038308)
- Cannot downgrade iD808s that support a configurable TFTP port to an earlier
version that does not support a configurable TFTP port (F0038311)
- When downgrading an iD808 from dual-stack to non-dual-stack, any statically
configured iCMS IP addresses will be lost (F0038330)
Known Defects/Issues in Version 3.532.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.531.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.531.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.531.1.0
- Improved internal DSP/Linux communications and diagnostics
Defects Resolved in Version 3.531.1.0
- No defects resolved in this release
Known Defects/Issues in Version 3.531.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.530.3.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.530.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.530.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.530.3.0
- Sendlogs doesn't work on the iD808 (F0038258)
Known Defects/Issues in Version 3.530.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.530.2.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.530.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.530.2.0
Defects Resolved in Version 3.530.2.0
- iDUCX does not report iCDS errors to the ARIA UI (F0038066)
- No audio heard on speaker channels or point-to-point intercom calls when using ARIA if the master
volume knob is turned down when ARIA connects to the iD808 (F0038095)
- NTP clock error when configured with a DNS host name (F0038221)
- The iD808 can crash if the user seated on the device has too many appearances configured (F0038224)
- If after requesting logs to be downloaded from a iD808 in iManager, you then request another download
before the first has finished being created and sent, then the second download request is ignored, as
expected, but further attempts to download from iManager will fail as well (F0038226)
- When configured for handset auto-clear and rapidly selecting different SbRTP MRD appearance keys,
the handset busy warning message is presented to the user (F0038227)
- When logging out, if there are MRD calls on speaker channels, the CDR CALL ENDED events are not sent out
for those calls (F0038241)
Known Defects/Issues in Version 3.530.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.530.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.530.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.530.1.0
- Swap keys menu option (swap any two paginating keys)
Defects Resolved in Version 3.530.1.0
- Ethernet ports go down on iE801 modules when both ports are enabled with STP enabled (F0038199)
Known Defects/Issues in Version 3.530.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.520.1.0
- SBCommon Version 5.1
- SBID808 Version 2.3
SIP Interface Versions in Version 3.520.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.520.1.0
- Apply Cisco configuration changes automatically
- Synchronized recording warning tone beeps
- Alert editing read-only override
- More tolerance to syncing to Window NTP servers
Defects Resolved in Version 3.520.1.0
- When the recording warning tone is enabled and a call is on handsfree, the transmit
recording warning tone is played at a fixed high volume level instead of the configured
user volume level for the recording warning tone (F0038161)
- The redundant failover mode for the ethernet ports on the iE801 does not work correctly (F0038163)
- MTFIF can crash if the SIP logs are enabled and left running for a long time (F0038171)
- The IDUCX devices report MCC as down when the only MCC messages being received are MCC OPEN messages,
which is the case when no calls are active (F0038178)
Known Defects/Issues in Version 3.520.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.510.4.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.510.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.510.4.0
- No features or enhancements added to this release
Defects Resolved in Version 3.510.4.0
- The iE801#1 IP Address and iE801#2 IP Address fields on the IP tab of the device in iManager is not updated with the iE801 modules IP addresses (F0038126)
- When DHCP for an iD808 is turned off in iManager, the iD808 reboots but does not have any IP address (F0038127)
- iGS CTI connection to iD808 is lost soon after being established (F0038128)
Known Defects/Issues in Version 3.510.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.510.3.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.510.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.510.3.0
- User based voice recording configuration supported
- Voice recording configuration status icon added to status bar
- Recording Configuration status screen added to Device Info
Defects Resolved in Version 3.510.3.0
- ARIA doesn't work when using TCP and the "Allow UDP SIP Signalling" option is unchecked on the RTP Media policy for the device (F0037624, F0037710)
- Occurrence of no audio between turrets when using Ethernet NIC resilience with STP disabled on the iD808 (F0038083)
- iD808 not accepting the CDR_SECONDARY_SERVER_ADDR CM variable (F0038103)
- Missed call indicator persistence after logon does not work (F0038112)
Known Defects/Issues in Version 3.510.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.510.2.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.510.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.510.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.510.2.0
- When set for CDR protocol version 8 some CDR call ended events can be generated which were only introduced in CDR version 9 (F0038067)
- Call packet information for RTP streams on iE801s not appearing when iD808 has two iE801s (F0038072)
Known Defects/Issues in Version 3.510.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.510.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.510.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.510.1.0
- Chinese (Simplified) language supported
- Maximum call register size increased to 100 entries per call register
- Persistence call register supported
- CDR protocol version 9 supported
- Call Info status screen enhanced to report 'Packets Received', 'Lost Packets' and 'Jitter'
- TFTP port configurable
- Geo failover enabled for iDUCX devices
Defects Resolved in Version 3.510.1.0
- With the Recording Tone enabled, a beep is generated regardless of whether the voice recording destination is configured or not (F0037790)
- A Cisco call stuck on the handset in the on-hold state and then in the busy elsewhere state after a failed privacy attempt (F0038045)
- When assigning an appearance to a paging speaker key, the appearance is cleared off any speaker keys on other speaker pages (F0037974)
Known Defects/Issues in Version 3.510.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.500.5.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.500.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.500.5.0
- No features or enhancements added to this release
Defects Resolved in Version 3.500.5.0
- Autodiscovery error message logged when sending a response back to the Autodiscovery request (F0037997)
Known Defects/Issues in Version 3.500.5.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.500.4.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.500.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.500.4.0
- No features or enhancements added to this release
Defects Resolved in Version 3.500.4.0
- Cisco common lamping not working correctly on iDUCX devices (F0037892)
- SSH on iDUCX does not connect (F0037916)
- The CM commands are not available under the SBEngineer account (F0037917)
- Paired muting / ganging does not work (F0037918)
- Turret not sending full iCMS error message to ARIA (F0037923)
Known Defects/Issues in Version 3.500.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.500.3.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.500.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.500.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.500.3.0
- Upgrading from an earlier version to v3.500 gives the device a new IP address as expected, but should release
the DHCP lease of the old address during the upgrade (F0037841)
- CDR connection does not work (F0037846)
- Ganging / Local Muting with paired device doesn't work after repowering (F0037857)
- The MCC is reported as 'Down' on iDUCX using Aria Touch (F0037890)
Known Defects/Issues in Version 3.500.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.500.2.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.500.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.500.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.500.2.0
- DTMF tones do not work on voicemail (F0037834)
- After a power cycle, the network icon shows yellow and there is no reason displayed in the show network menu (F0037836)
- Cannot configure an iCDS connection (F0037846)
- SbRTP ARD not working properly (F0037851)
- SbRTP Hoot voice service is being received on another channel (F0037858)
Known Defects/Issues in Version 3.500.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.500.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.500.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.500.1.0
- Dual IP stack (IPv6 ready)
- DSCP value for SIP messages configurable
Defects Resolved in Version 3.500.1.0
- No defects resolved in this release
Known Defects/Issues in Version 3.500.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.422.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.422.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.422.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.422.1.0
- user.err UI: **ERROR:UI_glib_log_handler:../src/UI_init.c: g_strstr_len: assertion `haystack != NULL'
failed errors in messages file when in a Cisco call (F0037678)
- **ERROR:vMtfifSubscribe:appl/mtfif/mtfif_subs.c: RvSipCallLegSetLocalAddress failed error logged when
logging in or resyncing a device that is registering to iCS using TCP SIP signalling (F0037700)
- ARIA isn't sent a call-ended message for the answerback group call when the other end responds and a
new point-to-point call is made (F0037719)
- Unattended transfer fails if the SIP 180 ringing messages is not received and only a SIP 200 OK is
received on the second leg of the transfer (F0037727)
Known Defects/Issues in Version 3.422.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.421.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.421.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.421.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.421.1.0
- Resource leak in the SIP stack when registering to a Cisco PBX using TCP. Once the pool of resources
are used up, this may result in call failures or common lamping errors until SIP registration is lost,
which will free up some resources (F0037671)
Known Defects/Issues in Version 3.421.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.420.4.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.420.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.420.4.0
- No features or enhancements added to this release
Defects Resolved in Version 3.420.4.0
- Common Lamping is not working for SbRTP MRD and DMVS MRD (F0037284)
- Call transfer issues on Aria for Cisco appearances (F0037328)
Known Defects/Issues in Version 3.420.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.420.3.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.420.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.420.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.420.3.0
- Incorrect / misleading error message when in a transfer (F0036798)
- When assigning an intercom appearance to a speaker channel on an iDUCX device the profile update
sent to iCMS is incorrect (F0036803)
- When receiving multiple point-to-point intercom calls it is possible for DspProc to be left with a
dangling call link that can result in future calls being established with no audio (F0037226)
- The manual device failover setting is accepted by the iDUCX but not the iGS resulting in ARIA sessions
failing to connect when the iDUCX devices are registered to the Secondary PBX (F0037235)
- A DSP link mismatch error may occur when running the mass conference call test on the regression suite (F0037240)
- Tiles are missing from ARIA Touch if the key KeyIndex for the tile is 64 or higher (F0037300)
Known Defects/Issues in Version 3.420.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.420.2.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.420.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.420.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.420.2.0
- An incoming call to Aria shows a dash instead of the Caller ID (F0037116)
Known Defects/Issues in Version 3.420.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.420.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.420.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.420.1.0
- Multiple streams support for Aria
Defects Resolved in Version 3.420.1.0
- No defects resolved in this release
Known Defects/Issues in Version 3.420.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.411.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.411.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.411.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.411.1.0
- The "From Display Name" being unchecked should be ignored for Intercom calls using the
iCS. Otherwise the calls are shown as "Unknown" when intercom privacy is enabled (F0037063)
- When voice recording is configured, SbRTP on receive stream 3 will be muted (F0037070)
Known Defects/Issues in Version 3.411.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.410.3.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.410.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.410.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.410.3.0
- The voice recording warning tone is not being sent to the voice recorder (F0036915)
- When powering up or upgrading iD808s with iE801 modules fitted, sometimes the interface to the DSP locks
up, causing issues such as the device reporting DHCP on the iE801s has failed, MCC is down and media for calls not working (F0036928)
- When you select a Mitel PBX and change the 'length' and 'prefix' under the 'Inbound' and 'Outbound' tabs
the turret should be able to make the changes without the iD808 flagging up a restart required message (F0036993)
Known Defects/Issues in Version 3.410.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.410.2.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.410.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.410.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.410.2.0
- Handset microphone icon may show muted when it is not muted on the Aria UI when assigning Voice Service on Speaker Channel to Handsets (F0036812)
- The microphone icon on the Aria UI may change from unmuted to muted when putting a voice service onto the handset (F0036878)
- 'DSP Link Mismatch Detected' on an iD808 with iE801 modules (F0036909)
- When testing the manual geographical failover feature, when the iCS failback happens and the primary PBX is restored
to active, the devices do not automatically register and the DMVS voice services and MCC stay down (F0036916)
- iCS geographical failover does not work correctly when iCS resilience is set up as well (F0036924)
- Endpoints have failed to register with the secondary PBX with manual geo failover and iCS HA enabled (F0036937)
- The CDR messages do not show the iD808 IP address in the iE801 address field when the iD808 is sourcing those streams (F0036947)
- In the iD808 iCMS status SNMP trap the text fields report 'Restart required' when a resync is required to apply a
configuration change. This should report 'Resync required'. (F0036982)
- The iD808 SNMP traps include null strings if there is nothing to report. These should be a single dash (F0036983)
Known Defects/Issues in Version 3.410.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.410.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.410.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.410.1.0
- All 7 voice recording streams sourced from the iD808 when no iE801 modules are fitted
- iD808 UI control of the Voice Recording Warning Tone may be disabled in iManager
- The Voice Recording Warning Tone may be individually enabled or disabled for different call types in iManager
- Manual geographic failover to backup iCS server supported
Defects Resolved in Version 3.410.1.0
- iCMS is not accepting an update from the iD808 of a voice service being placed on to a speaker channel (F0036870)
- Recording Warning tone occurring unexpectedly on hoots (F0036399)
Known Defects/Issues in Version 3.410.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.400.2.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.400.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.400.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.400.2.0
- Possible to lose part of a iD808 display by selecting an intercom appearance (F0036565)
- iD808 doesn't send a profile update to iCMS after wiping an intercom appearance from a
speaker channel in SoftClient mode (F0036692)
- When the secondary iCDS Server connection goes down, the SNMP trap reports "Ethernet port 2
is down" (F0036705)
- On Aria Touch you cannot make a call if you seize the line by pressing the line tile (F0036722)
- iCMS doesn`t echo the Aria touch webpage when the user on the Aria touch webpage removes an
assignment on a speaker channel (F0036735)
- An internal server error on Aria touch webpage when I enable the call forwarding (F0036737)
- There may be a "user busy" message when clearing a participant out of a group call (F0036779)
- The error dialog box shows "3800 not defined <i>i</i>Manager" when there is a PBX / SIP server error (F0036781)
Known Defects/Issues in Version 3.400.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.400.1.0
- SBCommon Version 5.1
- SBID808 Version 2.2
SIP Interface Versions in Version 3.400.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.400.1.0
- ARIA Touch support
- Point-to-point intercom and group call support for ARIA
- SNMP traps enhanced to have a single trap per status type and to improve the variable bindings associated with the traps
Defects Resolved in Version 3.400.1.0
- Sometimes a local device can get stuck in the connecting state when originating a call (F0036438)
- iD808 crash when using soft client and a clearing VAD indication is sent out (F0036456)
- Possible SNMP issue after an upgrade (F0036463)
- Local muting may not work when using iDUCX devices (F0036508)
- In SoftClient mode, the turret doesn't send a message to clear Mute Alerts Now when the call on handset
is assigned to a speaker channel (F0036547)
Known Defects/Issues in Version 3.400.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- Possible to lose part of a iD808 display by selecting an intercom appearance (F0036565)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.330.2.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.330.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.330.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.330.2.0
- When originating a voice service (e.g. group call) from a local device, it is established but sometimes no audio is heard on the receiving endpoints (F0036433)
- Sometimes, two SIP status icons may be shown on the main screen (F0036517)
- Simplex dipping / muting may not work (F0036525)
- Audio Restore Delay config changes made in iManager do not change the setting in the iD808 until after a re-sync (F0036532)
- The Speaker Key Priority menu setting is not consistent with the equivalent setting in iCMS (F0036546)
Known Defects/Issues in Version 3.330.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.330.1.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.330.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.330.1.0
- Local dipping
- Paired device for local dipping
Defects Resolved in Version 3.330.1.0
- When dialling a group call, the microphone is active before the call is connected (F0036493)
Known Defects/Issues in Version 3.330.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.320.6.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.320.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.320.6.0
- No features or enhancements added to this release
Defects Resolved in Version 3.320.6.0
- Conferences do not clear correctly off the handset for some PBX types. E.g. Mitel (F0036313)
- Aria sessions not showing incoming calls although you can hear ringing (iCS and Cisco calls) (F0036314)
- A UI process crash when processing SCMP message requestInfo with "callStatus" as argument. May
happen during ARIA logins (F0036318)
Known Defects/Issues in Version 3.320.6.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.320.5.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.320.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.320.5.0
- No features or enhancements added to this release
Defects Resolved in Version 3.320.5.0
- DSP Link Mismatch Detected on iD808 (F0036263)
- When a Cisco call with privacy on is put on hold, the padlock icon correctly changes from locked to
unlocked but instead of showing the called number the word 'Conference' is shown (F0036274)
Known Defects/Issues in Version 3.320.5.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.320.4.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.320.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.320.4.0
- No features or enhancements added to this release
Defects Resolved in Version 3.320.4.0
- When failing to subscribe to an iCS server, error logs can be generated which over time will fill the
log files (F0036226)
- When using G.729 for recording streams, SIP calls did not offer g.729 and the resources had been used
up by the recording streams (F0036240)
Known Defects/Issues in Version 3.320.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.320.3.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.320.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.320.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.320.3.0
- When you try to make a call to an inactive voice service from ARIA it's rejected by the session
controller (iD808) with an "Internal error". This should be rejected but with a better error message
because "internal error" is not a good user error. (F0036103)
- No response from session controller when a SoftClient ARIA sends a privacy request on a handset
when the handset is not in a call (F0036177)
- When a turret clears an iCS call belonging to a three-device conference the label still says
"Conference" on the other devices (F0036204)
Known Defects/Issues in Version 3.320.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.320.2.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.320.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.320.2.0
- TCP SIP signalling with a Cisco PBX
- Re-subscription to iCS halted when registration is lost
Defects Resolved in Version 3.320.2.0
- Delay in setting up media with Cisco CUCM v11.5(SU3) (F0035875, F0035911, F0035912)
- Wrong style is sent in the callNotify/callProgress message when an appearance key is configured with
the alert disabled (F0036071)
- The spelling of iCMS is wrong in the Device Info menu of the iD808 (F0036083)
- When using the iD808 with German Translations, translations for the Push-To-Answer settings are missing (F0036094)
- In Aria, ARD calls when on speaker with mic set to talk and barged in on another Aria session do not end correctly
when the clear button is pressed (F0036139)
- iD808 crash when stopping talk mode on speaker channel for remote group calls (F0036169)
Known Defects/Issues in Version 3.320.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.320.1.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.320.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.320.1.0
- Improved SbRTP diagnostics with a configurable SbRTP Inactivity timeout
- Enhancement to iCS registration functionality to allow the registration expiry, delay and timeout
to be configurable
Defects Resolved in Version 3.320.1.0
- When presenting an incoming call while the iD808 has screen saver active the handset LED becomes
inactive. On some endpoints the LED is not affected (F0036055)
Known Defects/Issues in Version 3.320.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.310.2.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.310.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.310.2.0
- No features or enhancements added to this release
Defects Resolved in Version 3.310.2.0
- iD808 is sending an empty profile update message to iCMS when it is powered on. This is in addition to
the expected announce message (F0035819)
- Privacy doesn`t work on iD808 against Cisco uCM v11.5(1)SU3 (F0035873)
- When logging out a user or resynchronising an iD808, the master iCS appearance may not be unsubscribed.
The log messages may be filled with "Dialog message empty" messages (F0035884)
- The "Description" field in the SNMP trap browser shows an iCMS profile error as an iD808 Network (F0035886)
- Using purely TCP SIP you cannot transfer an iCS telephony from one iD808 to another (F0035887)
- If all SNMP traps are triggered at the same time it is possible that one or more traps will not be sent (F0035900)
- After using conferencing and push to handset on the iD808 a hoot on a speaker channel was still attached
to the handset even though the UI showed the call disconnected (F0035903)
- When the SNMP example.conf file is modified via an upgrade, the iD808 may not restart the SNMP service and
therefore may use the old configuration until restarted (F0035906)
Known Defects/Issues in Version 3.310.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SNMP MIB Versions in Version 3.310.1.0
- SBCommon Version 5.0
- SBID808 Version 2.0
SIP Interface Versions in Version 3.310.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.310.1.0
- UDP/TCP SIP Signalling configuration
- UDP SIP Signalling can be disabled
- Cisco CUCM configuration retrieved via HTTP GET instead of TFTP
- SNMP enhanced to send traps using Speakerbus OIDs and include more details of fault conditions
Defects Resolved in Version 3.310.1.0
- The global muting signal may be sent for an outgoing call when early media is offered and this can
cause one-way voice (F0034823,F0035779)
Known Defects/Issues in Version 3.310.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.301.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.301.1.0
- SbRTP Inactivity Timeout configurable via iManager
- SbRTP dropouts stats added to DSP
Defects Resolved in Version 3.301.1.0
- No defects resolved in this release
Known Defects/Issues in Version 3.301.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.10.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.10.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.10.0
- When a Cisco shared call appearance is set to private on one user and taken off again, the other
shared appearance disappears from Aria and CTI (F0035775)
- When a Cisco shared call appearance is called and then the call is transferred to the user who
share's the call appearance, the iD808 crashes (F0035783)
- When trying to set a conference to private on Aria, the message 'internal session controller
error' is displayed instead of 'privacy unavailable in conference (F0035788)
Known Defects/Issues in Version 3.300.10.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.9.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.9.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.9.0
- On Aria, if a user drops out of a shared line that is connected to a barged-in conference on a speaker channel,
the appearance goes back to idle instead of busy-elsewhere (F0035745)
- When answering a Cisco call on speaker channel, the call is answered but the the speaker channel may continues
to show ringing (F0035753)
- In Aria (iD808 hosted and iCB), when there is a conference on the handset and transfer is press an 'internal
session controller error' is displayed instead of a normal rejection message (F0035758)
Known Defects/Issues in Version 3.300.9.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.8.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.8.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.8.0
- Using CTI, when transferring a call on an Avaya appearance to another user also signed into CTI,
the call doesn't appear answered (stuck in call state alerting) the call does however answer
correctly on the iD808 (F0035730)
- When using CTI, an MRD can disappear from the list of active calls when the incoming alerting is stopped (F0035737)
- Occasionally, when an Avaya call is transferred using CTI, two call lines are seen in CTI of
the receiving user. Also when doing the same thing with Aria logged in two call lines are also seen (F0035741)
- You are able to place multiple calls on a single speaker channel when using Aria (F0035743)
Known Defects/Issues in Version 3.300.8.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.7.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.7.0
- Call end may be incorrectly sent to CTI when using shared lines (F0035717)
Known Defects/Issues in Version 3.300.7.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.6.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.6.0
- "undefined" may be seen on the CTI page when a shared line is barged in to (F0035671)
- Issue with DSP Link mismatch reporting for P2P Intercom and Group calls put on hold (F0035687)
- Missing/incorrect label when editing speed dial originated from a group directory (F0035688)
- Invalid operation error displayed in ARIA when trying to answer a VPW call on a speaker (F0035700)
Known Defects/Issues in Version 3.300.6.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.5.0
- Remove security vulnerability CVE-2002-0510
Defects Resolved in Version 3.300.5.0
- The volume levels and muting status are not being reported correctly in ARIA after re-connecting (F0035472)
- With CTI, sometimes when moving a conference from a speaker channel to a handset no
audioDeviceStatus messages are sent to the iGS (F0035482)
- VPW needs a dial number to effectively work in CTI (F0035333)
- Hanging call in busy elsewhere state in CTI client after repetitive telephony calls (F0035532)
- All voice services, both DMVS and SbRTP do not work (F0035610)
- When triggering a live update that change talk permissions for SbRTP Hoots, two calls become
visible in CTI and it is out of sync with the turret (F0035628)
Known Defects/Issues in Version 3.300.5.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.4.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.4.0
- When a shared Cisco call appearance is set to private on one unit the label on the bridged appearance
shows 'conference' (F0035464)
- If you seize a line and dial without a number in CTI then on both Cisco and Avaya the line gets stuck
in 'calling out' mode for a few minutes (F0035470)
- Internal error received from the iD808 when assigning audio devices in ARIA (F0035478)
- When using CTI to place a call, the iD808 does not come out of screen-saver (F0035501)
- When logged into CTI the "CTI session connected" message is added to the log file every 30 seconds (F0035525)
- With auto-hide enabled and the menu displayed, if the menu is hidden by an incoming call and then
a remote ARIA session is started the hidden icon is still displayed, when it should be cleared (F0035527)
- An occurance of an issue where the status bar shows that the menu is hidden and pressing the OK key shows
the warning "Menu in use on handset 1/2), but selecting the other handset still doesn't allow the menu to be entered (F0035528)
Known Defects/Issues in Version 3.300.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- The volume levels and muting status are not being reported correctly in ARIA after re-connecting (F0035472)
- With CTI, sometimes when moving a conference from a speaker channel to a handset no
audioDeviceStatus messages are sent to the iGS (F0035482)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.300.3.0
- A remote device dials a group call to all remote devices. A local device dials an AB group call with higher
precedence and a remote device answers putting the two in a point to point. Once the point to point has ended,
the group call sitting underneath the remote turret is stuck on the intercom line but you cannot get the group
call up on your screen (F0035285)
- A remote device dials a group call to all remote devices. A local device dials an AB group call with higher
precedence and a remote device answers putting the two in a point to point. Once the point to point has ended,
the group call stacked underneath the remote turret clears (F0035286)
- With DMVS MRD channels on speaker, when the iCS is restarted an additional SIP session may be created.
This may eventually slow down or stop the device or iCS (F0035302, F0035497)
- A turret may get into a state where no audio or ringing is heard on the turret (F0035334)
- An iD808 shows no logging is enabled after it`s repowered, but ICMS reports that the logging is active (F0035335)
- DSP link mismatch detected message seen on a remote iD808 device (F0035350)
- iDUCX should not request a resync when a user is being unseated (F0035378)
- Issue with CTI when using shared lines (F0035430)
- Issue with CTI when clearing a call that is on hold (F0035435)
- In ARIA, on a local device when a DMVS ARD is ringing and answered on another device the state is not
changed and continues to show ringing call (F0035445)
- When seizing a Mitel appearance using CTI, the label of the appearance is missing on call label (F0035452)
- There is a memory leak in MTFIF (F0035465)
- Incoming DMVS ARD calls can show the "VS_GW_x" label on the second line in two line mode (F0035483)
- On Cisco, when privacy is enabled, the caller ID briefly changes to "Caller ID Withheld" before changing
back to the correct number (F0035484)
- iD808 crash when logging into CTI with UI basic call logs enabled (F0035519)
Known Defects/Issues in Version 3.300.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- When a shared Cisco call appearance is set to private on one unit the label on the bridged appearance
shows 'conference' (F0035464)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.2.0
- Improvements to the CTI interface
Defects Resolved in Version 3.300.2.0
- No defects resolved in this release
Known Defects/Issues in Version 3.300.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- With DMVS MRD channels on speaker, when the iCS is restarted an additional SIP session may be created.
This may eventually slow down or stop the device or iCS (F0035302, F0035497)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.300.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.300.1.0
- Support for CTI via the iGS
- Screen saver day/night mode
- Support for bulk registering to the iCS with a master appearance
- Support for failover to a backup iCS server
- Caller ID configuration options to ignore P_Asserted-Identity and From-Display-Name headers
Defects Resolved in Version 3.300.1.0
- An incoming intercom P2P call with alerting turned off may show the splash screen over the top of the screen saver (F0035205)
- With the intercom audio device set to handsfree, and dialling a point-to-point intercom from the call register to a user
with intercom privacy enabled, the handsfree indicator is not illuminated until the call is answered (F0035208)
- iD808 devices may send registration messages to iCS when they have intercom only appearances and are remote devices (F0035348)
Known Defects/Issues in Version 3.300.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- With the intercom audio device set to handsfree, and dialling a point-to-point intercom from the call register
to a user with intercom privacy enabled, the handsfree indicator is not illuminated until the call is answered (F0035208)
- With DMVS MRD channels on speaker, when the iCS is restarted an additional SIP session may be created.
This may eventually slow down or stop the device or iCS (F0035302, F0035497)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.8.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.8.0
- No features or enhancements added to this release
Defects Resolved in Version 3.200.8.0
- With DMVS MRD channels on speaker, when the iCS is restarted an additional SIP session may be created.
This may eventually slow down or stop the device or iCS (F0035302, F0035497)
Known Defects/Issues in Version 3.200.8.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- An incoming intercom P2P call with alerting turned off may show the splash screen over the top of the screen saver (F0035205)
- With the intercom audio device set to handsfree, and dialling a point-to-point intercom from the call register
to a user with intercom privacy enabled, the handsfree indicator is not illuminated until the call is answered (F0035208)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.7.0
- No features or enhancements added to this release
Defects Resolved in Version 3.200.7.0
- After stacking group calls on a remote iD808 a 'DSP Mismatch Detected!' may be reported under the network status.
After the group calls have been ended the reported error is cleared (F0035189)
- A DMVS hoot on a speaker channel on iD808 may not work after doing a "Take node offline" test on a resilient pair of iCS servers (F0035190)
- The announce tone might not be heard when initiating a group call using DMVS (F0035194)
- A DMVS MRD on a speaker channel in the busy elsewhere state doesn't show the VAD indicator but audio is heard (F0035201)
Known Defects/Issues in Version 3.200.7.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- An incoming intercom P2P call with alerting turned off may show the splash screen over the top of the screen saver (F0035205)
- With the intercom audio device set to handsfree, and dialling a point-to-point intercom from the call register
to a user with intercom privacy enabled, the handsfree indicator is not illuminated until the call is answered (F0035208)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.6.0
- No features or enhancements added to this release
Defects Resolved in Version 3.200.6.0
- When making an outgoing group call attached to a speaker channel the VAD stays red with no audio being heard (F0034460)
- While the originator of an answerback group call is in the 'Press * to answer' state, if a point-to-point intercom
call is received and rings, and then the user presses * and the answerback call is answered, both calls drop (F0035048)
- If the turret is in the middle of dialling an intercom number on its second appearance and receives a group call which
it chooses to decline, the screen will freeze and you will be unable to enter digits until you re-seize the intercom line (F0035056)
- When moving a Voice Service from the handset to a speaker key in ARIA, the drop-down list that appears after you press
assign does not show the name of any VS that has already been put on a speaker key. All you see is a pair of empty brackets (F0035064)
- Precedence level issues are being seen when "P2P / DConf has highest priority" in Supervisor is ticked (F0035076)
- Cannot configure the SNMP managers (under the SNMP policy) separately. They all have to be the same IP address or SNMP breaks (F0035106)
- The speakerConfig messages sent to ARIA when the turret is in soft client mode, does not send "t-9/1" or "t-9/2" in the
appId for speakers that are linked to an intercom appearance. Instead it send "t4294967287/1" (F0035107)
- Speed dials setup to use corporate directory entries in iDUCX devices, do not get updated if the corporate directory is
updated when a user is seated (F0035145)
Known Defects/Issues in Version 3.200.6.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.5.0
- Speakerbus enterprise MIB supported
- SNMP traps and variables added for network, iCMS and SIP status
- Number of SNMP managers supported increased to 10
Defects Resolved in Version 3.200.5.0
- Incorrect reporting of a DSP link mismatch error (F0034976)
Known Defects/Issues in Version 3.200.5.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.4.0
- Re-invite timer for detecting network issues on all iCS call types
- DSP RTP link mismatch detection
Defects Resolved in Version 3.200.4.0
- The missed call register sometimes doesn't show the most recent entry (F0034335)
- A dangling RTP link may be left open after a network problem, which may result in one-way audio for a new call (F0034608)
- After importing, deleting and editing entries and sub-entries in the personal directory, can end up with the directory on the
iD808 not matching that seen in iCMS for the user seated at the iD808 (F0034735)
- When in loop-through mode on the Ethernet interfaces the BPDU packets are not passed through the iD808 (F0034815)
- Global muting may not work when the iCS telephony appearance is on a fixed key (F0034820)
- With an intercom appearance on a speaker channel and redialling a group call from the call register on the intercom splash
screen, the call state icon on the speaker channel is left in the connecting state (F0034839)
- When a number is redialled, the entries count in the XML file is decremented which may result in the redial/placed call
register appearing to be empty or exceeding the 40 entries maximum limit (F0034852)
Known Defects/Issues in Version 3.200.4.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.3.0
- Personal directory limit increased from 500 to 1000 (sub-entry limit remains at 5000)
- Detail View of the Call Register enhanced to display the far end number
- Push-to-answer / Toggle-to-answer enhanced to seize the default appearance when there are no calls to answer or clear
- TCP port 6000 not opened by X server
- Configuration option to ignore the Remote Party ID label in the SIP header
- Manual dial autoscripts now supported on iDUCX devices
- makevs added to autoscript commands
- Changes to the group call handling to match iCS version 2.500.6.0 changes
Defects Resolved in Version 3.200.3.0
- IGMP messages can use an incorrect source IP address (F0034568)
Known Defects/Issues in Version 3.200.3.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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From v3.200.2.0 onwards, any firmware downgrade to an earlier version must use the full installation process to
ensure the updated security libraries are correctly replaced by the older versions.
SIP Interface Versions in Version 3.200.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.2.0
- OpenSSL library updated to version 1.0.2k
- OpenSSH library updated to version 7.4p1
- Ignore handset status for group talk supported
- Long alerting indication supported
- Long on-hold indication supported
- Additional toggle-to-answer and push-to-answer handset modes supported
Defects Resolved in Version 3.200.2.0
- When establishing an answerback P2P call on a remote device with one previous (less important)
call on the stack, the call is not properly cleared if assigned to the speaker channel (F0034268)
- Audio is still being heard for remote group calls that are being replaced onscreen by a more
important call. The background call as well as the current call is played on speaker (F0034269)
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
- iD808 DMVS intercom splash screen doesn't clear when call is ended from local/far end iD808 (F0034359, F0034374)
Known Defects/Issues in Version 3.200.2.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
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SIP Interface Versions in Version 3.200.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.200.1.0
- DMVS intercom and group calls
- Toggle-to-answer handset mode added
- ‘Speaker channel VAD logging’ added to support regression testing
Defects Resolved in Version 3.200.1.0
- Privacy does not work when using Cisco UCM 11.5 (F0033324)
- When logged in to the iD808, and dialling from the directory the key will show the directory
label (e.g. John Smith). When using the ARIA web interface with the iD808 dialling the same number
through the directory the label is shown without the space (e.g. JohnSmith) (F0033976)
- Mute alerts on ARIA stops working when a call on handset 2 is ended (F0034024)
- When assigning an iCS telephone call, with global muting enabled, from a handset to a speaker channel
that is part of a group talk that is latched on, the global muting signal is not sent to line (F0034137)
- Call forwarding on Cisco fails to be disabled if your iCS intercom number is the same as your dial
number on Cisco UCM (F0034158)
- When using Push-To-Answer to answer call on lines that have different priorities, the priority is
ignored and the longest active call is answered (F0034169)
- If an iD808 has an intercom call, and it is on the splash screen, and the redial button is pressed, the
redial menu appears. The back button then goes back to the home screen instead of the intercom screen (F0034180)
- CDR talk state reporting when changing talk rights with live update (F0034234)
Known Defects/Issues in Version 3.200.1.0
- If the recording stream policy is changed while calls are active a RECORDING_STREAM_INFO event is not
generated. The information is correct the next time a call/heartbeat event is generated but can be
misleading at the point the recording stream info is changed (F0034061)
- When establishing an answerback P2P call on a remote device with one previous (less important)
call on the stack, the call is not properly cleared if assigned to the speaker channel (F0034268)
- Audio is still being heard for remote group calls that are being replaced onscreen by a more
important call. The background call as well as the current call is played on speaker (F0034269)
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.101.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.101.2.0
- Instant Playback user interface enhanced
Defects Resolved in Version 3.101.2.0
- No defects resolved in this release
Known Defects/Issues in Version 3.101.2.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.101.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.101.1.0
- No features or enhancements added to this release
Defects Resolved in Version 3.101.1.0
- iD808 crash related to the call register (F0034038)
Known Defects/Issues in Version 3.101.1.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.100.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.100.7.0
- Instant Speaker Playback enhanced to improve the warning message shown when there is no audio to playback
- Japanese and German translation files updated for new features
Defects Resolved in Version 3.100.7.0
- An iCS missed call may not be logged to the call register (F0033987)
Known Defects/Issues in Version 3.100.7.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.100.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.100.6.0
- Instant Speaker Playback enhanced to skip over leading silence
Defects Resolved in Version 3.100.6.0
- On ARIA, when a call with constant VAD activity is assigned to a speaker channel, the VAD is
shown as off until the VAD goes off and back on again (F0033957)
Known Defects/Issues in Version 3.100.6.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.100.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.100.5.0
- No features or enhancements added to this release
Defects Resolved in Version 3.100.5.0
- A DMVS ARD barge-in attempt from a ‘local’ unit for a call at the talker limit are sometimes
incorrectly connected by the local SE708 (F0033142)
- An SbRTP MRD may get stuck in the "attempting to listen" mode (F0033531)
- Message protocol error seen when trying to clear a DMVS hoot from the handset on Aria,
the way we got rid of it was to reset the session (F0033829)
- Can get in a situation where the user cannot clear down a DMVS hoot from the handset,
producing a message protocol error (F0033854)
- Master mic mute doesn't work after moving an ICS or Mitel call from speaker channel
to handset and then to another speaker channel (F0033910)
- The "Add" lines menu option is always available and shows random information if no
lines can be added to keys (F0033912)
- An instance of the handset remaining connected to a hoot after the hoot was cleared from the
handset. This meant audio was heard at other users on the hoot without the talker realising.
Synchronising the iD808 cleared the issue (F0033916)
- An occurance of a UI locked up while logging into an ARIA session (F0033924)
- No label shown for a DMVS MRD in the busy-elsewhere state on the Call Activity panel in ARIA (F0033925)
- iDUCX devices report "DSP_ProcessMessageAck:../src/dsp_control.c: DSP command failed response
========>>>>>> 183 MESSAGE_NOT_IMPLEMENTED" in the messages log during start-up (F0033927)
- German labels for intercom group calls on the iD808 are truncated in the intercom splash
screen and are not totally visible for the user (F0033933)
Known Defects/Issues in Version 3.100.5.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.100.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.100.4.0
- No features or enhancements added to this release
Defects Resolved in Version 3.100.4.0
- When using a Speaker Channel mic for the first time the icon is grey (not red or green).
Selecting it at this point there is a Invalid Operation drop down displayed (F0033515)
- An issue with max talker limit on SbRTP hoot on a speaker (F0033830)
- An SbRTP MRD may get stuck in the "attempting to listen" mode (F0033531)
- iTurret in a crashed/locked state after changes made to the Cisco switch ports (F0033846)
- DMVS ARD stays ringing at far end after ARD is cleared down, reset of Aria session clears the
ringing call (F0033855)
- Cisco registration fails if the “Cisco Device Name Prefix” is different from the iTurret
“Logon Name” (F0033877)
- When using an iDUDX for a speaker channel hoot, the microphone sometimes will not change
to mute and stays open (F0033879)
Known Defects/Issues in Version 3.100.4.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.100.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.100.3.0
- Support added for the iCMS 'Cisco Device Name Prefix' configuration
Defects Resolved in Version 3.100.3.0
- iD808 sends an empty profile to iCMS when an ARIA session connects to it. If the iD808 can't
contact the iCMS at this point, it will go out-of-sync (F0033579)
- The iD808 Call Information does not show the codec information until the call is shown as connected
even though we could hear that the call had connected to the voice mail server (F0033718)
- With "auto handset mute" enabled, when in an ARIA session it is possible to get to a state
where transmit audio is muted but not shown as muted on the ARIA UI (F0033757)
- With DND set to ON whilst Aria is logged in, if the user logs off, the icon is lit on the
iTurret, but the function key shows DND as disabled (F0033781)
- When seating a user with intercom but no iTurret credentials on an iTurret , the iCMS icon on
the iTurret is yellow. The iCMS Status in Device Info states there is a profile error and that
the default appearance is not configured (F0033792)
- When disabling master volume control the volume levels are sometimes incorrect (F0033802)
- During instant playback the handset volume control does not work properly (using the volume
control soft key). This is heard when the call is displayed on the iD808 and not iE801 (F0033803)
- After an iDUCX device is created and synchronised from iCMS it is possible that the configuration
for the keys are lost (F0033813)
- When starting up the iDUCX containers sometimes the first announce to iCMS is sent out with a
message ID greater than 1 which results in the iDUCX device being out of sync (F0033815)
- An iDUCX "crashed" whilst making a simple Cisco P2P SIP call whilst Aria session connected (F0033819)
- Alert profile with VPW set to ring once doesn't ring. Change it to ring continually and it works
correctly (F0033843)
Known Defects/Issues in Version 3.100.3.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.100.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.100.2.0
- Instant Speaker Playback
- Supports an alternative Ethernet Resilience mode that does not use STP
Defects Resolved in Version 3.100.2.0
- Display problems when barging into Cisco v11 calls from a ARIA session on an iDUCX device (F0033523)
- Using iDUCX, there is no audio on SbRTP hoots (F0033570)
- The advanced logging options (SIP, other, log filtering) on iDUCX devices do not work (F0033735)
- When the sip stack in iDUCX reserves a complex codec, that complex codec stays reserved regardless of
whether it is used by RTP connection. Thus all SIP calls will reserve a complex codec whether or
not it is used. This is wastefull of resource (F0033758)
Known Defects/Issues in Version 3.100.2.0
- With "auto handset mute" enabled, when in an ARIA session it is possible to get to a state
where transmit audio is muted but not shown as muted on the ARIA UI (F0033757)
- With DND set to ON whilst Aria is logged in, if the user logs off, the icon is lit on the
iTurret, but the function key shows DND as disabled (F0033781)
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.9.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.9.0
- No features or enhancements added to this release
Defects Resolved in Version 3.000.9.0
- With "auto handset mute" enabled, when in an ARIA session it is possible to get to a state
where transmit audio is muted but not shown as muted on the ARIA UI (F0033757)
- With DND set to ON whilst Aria is logged in, if the user logs off, the icon is lit on the
iTurret, but the function key shows DND as disabled (F0033781)
Known Defects/Issues in Version 3.000.9.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.8.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.8.0
- No features or enhancements added to this release
Defects Resolved in Version 3.000.8.0
- Occasional a SbRTP hoot on an iD808 speaker isn`t active (no triangle or circle) (F0032882)
- Possible iD808 UI crash when editing a speed dial key (F0033775)
Known Defects/Issues in Version 3.000.8.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.7.0
- No features or enhancements added to this release
Defects Resolved in Version 3.000.7.0
- A DMVS ARD gets into a state (ringing / not ringing) when enabled from iWS with no iG330 present (F0033088, F0033354)
- In ARIA, cannot take a Mitel call appearance off hold (F0033462)
- DMVS ARD legend is blank when bridged to an iD808 signed into iWS. It shows just a dash (no information) in the
iWS call activity monitor. (F0033700)
Known Defects/Issues in Version 3.000.7.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.6.0
- No features or enhancements added to this release
Defects Resolved in Version 3.000.6.0
- An occurance of an appearance being shown as busy elsewhere, when it hadn`t been used (F0033434)
- Unattended transfer for a Mitel call with a speaker key assigned causes the call to transfer correctly
but leaves the speaker key appearing active (F0033560)
- Issues with failed transfers on Cisco CuCM v11 on both CloudFront and CloudBase (F0033578)
- Pressing speaker with SbRTP hoot assigned, the first time it stays in "attempting to connect" state.
Disable and press again and the hoot connects and works (F0033595)
- Session Controller Icon in Aria is red (F0033630)
- When using iE801s and the Master Volume is enabled for the unit there is a problem when changing from
split to dual speaker mode. The level is set to unity gain instead of that of the volume control (F0033634)
- On ARIA, the handset can end up in the wrong state when attempting to move a seize line to a busy
speaker channel (F0033644)
- Remote Syslog messages from the iDUCX devices report the device type as iD808 and not iDUCX (F0033648)
Known Defects/Issues in Version 3.000.6.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.5.0
- Only iCB server features in this release. See iCB version 1.000.11.0.
Defects Resolved in Version 3.000.5.0
- When I barge into a P2P call on Avaya Aura, speak into the barged in conference for a few seconds,
then barge out, the CONFERENCE legend takes up to 65 seconds to clear down (F0033521)
- Sending a partial profile update to the iD808 (or iDUCx) whilst they are being used by a Soft Client,
results in them sending an invalid XML response back to the CMS in response to the update from the CMS (F0033533)
Known Defects/Issues in Version 3.000.5.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.4.0
- Only iCB server features in this release. See iCB version 1.000.10.0.
Defects Resolved in Version 3.000.4.0
- No defects resolved in this release
Known Defects/Issues in Version 3.000.4.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.3.0
- No features or enhancements added to this release
Defects Resolved in Version 3.000.3.0
- If you press the mute all button on an iD808 and then take the iD808 over with an ARIA web application
to put it in soft client mode, the mute all red banner is still shown (in soft client debug mode)
but the channels are not actually muted (correctly) and when you exit soft client mode the mute all
red banner is still there but the channels are not actually muted (F0033101)
- Soft client speaker VAD indication may be left on when the other end of a call is taken off hold (F0033163)
- It is possible to connect an intercom call from an iD808 to an Aria session (connected to an iD808)
with audio present (F0033289)
- The title bar legend does not echo what has been set for the default appearance for the user whilst
the Aria session is connected (F0033292)
- Missed call log doesn`t show up a missed call (F0033296)
- "DSP command failed response ========>>>>>> 19 MIXER_LEVEL_MIXER_MEMBER_NOT_FOUND" displayed in the
messages log when receiving a forced group call (F0033302)
- Incoming call, dynamic key LED does not flash when 'Screen Saver Auto-Exit timer' is set to [off]
when in screen saver (F0033335)
Known Defects/Issues in Version 3.000.3.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.2.0
- Soft client functionality enhanced to support iWS v1.000.12.0
- The logs' folder in sendlogs renamed to match the tar file name
Defects Resolved in Version 3.000.2.0
- When an iCS shared line is answered it may be listed as a missed call by other users (F0029297)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority incoming
group call stacked (F0032078)
- iD808 crash when migrating from one ICS to another (set up from iCMS) (F0032821)
- After a live update is received to add permissions to a voice service appearance the user starts
bulk registering to iCS using the intercom appearance as well as the call appearance (F0032824)
- SBEngineer doesn`t have access to the tail -f /var/log/messages command (F0032830)
- When 2 people put a SbRTP ARD on hold at the same time or nearly the same time, the call drops (F0032853)
- If the recordable attribute of a DMVS call in the talking state changes from OFF to ON, no change
in the CDR events is seen (F0032854)
- When a ringing ARD SbRTP call that take the device out of screen saver is answered by another user
the device goes not re-enter screen saver mode (F0032956)
- The group talk indicator on an iE801 remains on after a resync (F0032965)
- If a soft client session is started immediately after a resync or login, and before all registrations
and subscriptions have completed for the device, the ARIA UI will show that the device has SIP errors
and this error warning might not be cleared when eventually all registrations and subscriptions are
successful (F0032969)
- When using group talk it is possible for some channels to not clear down fully resulting in "crossed lines"
occurring (F0032974)
- A spelling mistake is in the iD808 online help (F0033029)
- Handset muting issue when using group talk (F0033030)
- Clear down tone continues to be heard after last call is cleared down by far end, when assigned
to speaker channel (F0033062)
- Resetting an ARIA remote session to an iD808 may take longer than necessary to connect (F0033189)
- SoftClient is receiving speakerConfig messages with cType="update" when requesting the speakerConfig
(either at initialisation or when changing speaker page). It should only send speakerConfig messages
with cType filled in when assigning from handset to speaker or when wiping speakers. (F0033215)
Known Defects/Issues in Version 3.000.2.0
- Soft client speaker VAD indication may be left on when the other end of a call is taken off hold (F0033163)
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 3.000.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 3.000.1.0
- The alert volume levels have been increased when in an ARIA session
Defects Resolved in Version 3.000.1.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- With an intercom appearance on a speaker channel, received answer back group calls cause
an informational popup to be displayed which blocks the view of the normal intercom splash
screen if an intercom call is received (F0032287)
- With a user with non-paginating keys configured on iE816 modules is seated to the vturret
there may be warnings about the lack of float key (F0032330)
- Crash / registration failure caused by a memory leak that occurred during iCS capacity
testing (F0032643)
- Muting issue on speaker channels when in an Aria session (F0032687)
- One off device lock up (F0032688)
- SBEngineer doesn`t have permissions to the autoscripts on the vturret (F0032717)
- 'CM tftpscript' fails on a vTurret (F0032718)
- UI crash in vUi_handle_subsn_result() (F0032721)
- After a live update which changes the available appearances to a user, the iD808 can end
up repeatedly trying to unregister a voice service to iCS, which is constantly rejected
by iCS as registration attempts are only permitted using call appearances or intercom
appearances (F0032786)
Known Defects/Issues in Version 3.000.1.0
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial,
directory dial, VPW or call register dial (F0034348)
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SIP Interface Versions in Version 2.902.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.902.2.0
- No features or enhancements added to this release
Defects Resolved in Version 2.902.2.0
- DSP link mismatch detection errors
Known Defects/Issues in Version 2.902.2.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.902.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.902.1.0
- Re-invite timer for detecting network issues on all iCS call types
- DSP RTP link mismatch detection
Defects Resolved in Version 2.902.1.0
- A dangling RTP link may be left open after a network problem, which may result in one-way audio for a new call (F0034608)
- Global muting may not work when the iCS telephony appearance is on a fixed key (F0034820)
Known Defects/Issues in Version 2.902.1.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.901.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.901.1.0
- No features or enhancements added to this release
Defects Resolved in Version 2.901.1.0
- Global muting may not work for iCS telephony calls when the call is originated from a speed dial, directory dial, VPW or call register dial (F0034348)
Known Defects/Issues in Version 2.901.1.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.11.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.11.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.11.0
- Clear down tone continues to be heard after last call is cleared down by far end, when assigned to speaker channel (F0033062)
Known Defects/Issues in Version 2.900.11.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.10.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.10.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.10.0
- Handset muting issue when using group talk (F0033030)
Known Defects/Issues in Version 2.900.10.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.9.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.9.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.9.0
- When using group talk it is possible for some channels to not clear down fully resulting in "crossed lines" occurring (F0032974)
Known Defects/Issues in Version 2.900.9.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.8.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.8.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.8.0
- When 2 people put a SbRTP ARD on hold at the same time or nearly the same time, the call drops (F0032853)
Known Defects/Issues in Version 2.900.8.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.7.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.7.0
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- Enabling network trace in iCMS for a vTurret doesn`t work (F0032363)
- On a soft client session, the microphone is shown as muted when it's active on a speaker
channel when barged into a call on a speaker (F0032422)
- Using soft client if you try to unmute (talk) on a speaker that is associated with an ARD
and that ARD is currently in far end ringing state the iD808 rejects the unmute request
on the speaker with an "Invalid operation" command rejected message (F0032481)
- SNMP doesn`t work on the vTurret (F0032485)
- Using the SoftClient interface, after transferring a speaker channel call to a handset,
if the speaker channel is muted and the call returned to the speaker channel, the received
audio can still be heard (F0032498)
- Listening volume is too low when using the soft client web interface (F0032582)
- The volume levels are different between handsets and speaker channels in soft client mode (F0032583)
- Debug message at error level generated by UI_get_call_log_entry() when in soft client mode (F0032584)
- Associated CM variables not updated when changing volume levels from the soft client interface
and hence they will be forgotten with a resync (F0032585)
- Volume levels changed to 7 when exiting soft client mode (F0032586)
- When logging into vTurret the SIP Status may show an errors for any unconfigured
expansion and privacy devices for the Cisco PBX registration (F0032588)
- When starting a soft client session, the handset volume may not reflect the UI reported setting (F0032589)
- Port funnelling for Cisco PBX not implemented in vTurret (F0032602)
- When a call is answered on a speaker channel the LED indicator shows talk as active but
the transmit audio does not work (F0032603)
- Remote syslog doesn`t work on the vTurret (F0032605)
- When an iCS VPW is answered on a soft client web interface it doesn't stop ringing on
the web interface (F0032612)
- Issue with call register redial in soft client caused by applying the outbound dial rules
to a manual dial or placed call registry dial. Fix requires ARIA web application 1.100.6.0 or later (F0032622)
- On a soft client call, the handset audio device may be showed as microphone muted,
but the microphone may not be muted (F0032623)
- After a failed firmware upgrade (because the files were not in the correct location)
the iD808 can no longer be contacted by iCMS. This can result in the user being seated
at two devices because iCMS can't log off the user at the iD808. Also the iD808
even after synchronisation still shows the failed firmware status (F0032637)
- The MAC address seen in the CDR events of the vTurret is incorrectly formatted (F0032640)
Known Defects/Issues in Version 2.900.7.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.6.0
- The wait period before removing the screen saver in response to an incoming call is now configurable
- LED scheme 2 enhanced to display busy-elsewhere with global muting signalling
- Soft Client support for call register
Defects Resolved in Version 2.900.6.0
- For an iCS shared line on speaker for multiple users if you release the speaker channel whilst talking,
you can hear a loud burst of audio and the VAD will indicate on a speaker channel of the other users
when they have their mic muted and Global muting is enabled (F0032307)
- Unable to dial voice mail on a seized line when in soft client mode (F0032344)
- When pressing the privacy button on Aria webpage the error
'UI_get_CtiCallState:../src/UI_cti.c: iSCMPCallstate=10 not handled' is logged in the
messages file (F0032348)
- When a key is being held down to send a DTMF digit and the menu is accessed, releasing the key
doesn't stop the tone (F0032416)
- If in a menu and an inbound intercom call displays the intercom splash screen, when the intercom
call clears and the intercom splash screen is removed the revealed menu has lines over it (F0032456)
- When calling a DMVS ARD from another unit, a receiving Soft Client displays a blank string for the call
in the call activity panel (F0032459)
Known Defects/Issues in Version 2.900.6.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.5.0
- Autoscript enhanced to support 'makecall', 'clearcall' and 'bargein' commands
Defects Resolved in Version 2.900.5.0
- CDR traffic not sent from a vTurret device (F0032279)
Known Defects/Issues in Version 2.900.5.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- With an intercom appearance on a speaker channel, received answer back group calls cause an
informational popup to be displayed which blocks the view of the normal intercom splash screen
if an intercom call is received (F0032287)
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SIP Interface Versions in Version 2.900.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.4.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.4.0
- Logon menu not shown when logging on to SoftClient if the menu is already displayed (F0032005,F0032202,F0032248)
- In SoftClient mode, try to generate a DTMF tone on a handset from the web page when there is
a call on both handsets fails with "Unsupported operation" (F0032107)
- Calls are not recorded when using the soft client (F0032139)
- When in soft client mode and with an active call on the handset, the screen saver never activates (F0032230)
- Soft client web page is showing the wrong caller ID for a direct dialled number (F0032245)
- When upgrading the SERVICE_SSH setting is not remembered and SSH will always end up enabled (F0032262)
Known Defects/Issues in Version 2.900.4.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
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SIP Interface Versions in Version 2.900.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.3.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.3.0
- Avaya inbound call to a bridge line continues to ring after being answered by an Avaya phone using SIPS (F0031643)
- Updates made to an iD808 layout in iCMS while in a soft client session do not get updated to the iD808
after logging out of the soft client session (F0032031)
- Call forward no answer leaves a call ringing on the call activity panel (F0032047)
- Can get the privacy cog in a waiting to enable privacy state (F0032049)
- Live update of a hoot on a speaker channel may be lost when in a soft client session (F0032050)
- If you have an alerting ARD shown in Soft Client and you change the speaker page selected in Soft
Client, the alerting ARD is updated from the iD808 with a call state of busy elsewhere (F0032052)
- A tone-to line speed dial cannot be used to dial a call if it contains a 'P' character (F0032061)
- Whilst in Aria mode, if you do a sync from iCMS, there are issues (F0032065)
- Issues caused by pressing privacy when accessing Avaya voicemail (F0032066)
- When dialling a call from the corporate directory on the SoftClient the outbound dialling rules are not applied (F0032072)
- Audible alerting may be heard when the only alerting line has the alert mode set to off (F0032089)
- A soft client without any speaker channels is getting additional invalid VAD messages for audio device
sp16 when in a call on a handset (F0032106)
Known Defects/Issues in Version 2.900.3.0
- DTMF tone can get stuck playing on the speaker when using an iE801 module (F0032060)
- In Aria, issues seen during a barge-in conferencing test (F0032068)
- iD808 drops an initiated answerback intercom p2p call, if there is another low priority
incoming group call stacked (F0032078)
- In SoftClient mode, try to generate a DTMF tone on a handset from the web page when there is
a call on both handsets fails with "Unsupported operation" (F0032107)
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SIP Interface Versions in Version 2.900.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.2.0
- No features or enhancements added to this release
Defects Resolved in Version 2.900.2.0
- If an iD808 is in soft client mode and a new soft client in another browser connects to it,
the existing session is disconnected but the new session never connects (F0032030)
- A crash seen after pressing RESET in the Aria window (F0032036)
- In Aria, when selecting call forwarding and enabling ALL the call forwarding buttons and then
setting them to off, then pressing OK, Messaging Protocol error is displayed (F0032037)
- When clearing voicemail on Aria, it takes up to 2 minutes for the voicemail icon to go out (F0032039)
Known Defects/Issues in Version 2.900.2.0
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SIP Interface Versions in Version 2.900.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.900.1.0
- Soft Client support added for Line Seize, Voice Mail, Mute Alerts Now and Tone-to-line Speed Dial
- Soft Client startup handshake protocol supported
Defects Resolved in Version 2.900.1.0
- When a Soft Client session is ended on an iD808 and control returns back to the actual hardware
iTurret, the iCMS Status page states that the unit is in a logged out state when it is not (F0031915)
Known Defects/Issues in Version 2.900.1.0
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SIP Interface Versions in Version 2.800.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.800.3.0
- No features or enhancements added to this release
Defects Resolved in Version 2.800.3.0
- Avaya inbound call to a bridge line continues to ring after being answered by an Avaya phone using SIPS (F0031643)
- When a Soft Client session is ended on an iD808 and control returns back to the actual hardware iTurret, the iCMS Status
page states that the unit is in a logged out state when it is not (F0031915)
- Audible alerting may be heard when the only alerting line has the alert mode set to off (F0032089)
Known Defects/Issues in Version 2.800.3.0
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SIP Interface Versions in Version 2.800.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.800.2.0
- Logon menu added to regain control of a turret in Soft Client mode
Defects Resolved in Version 2.800.2.0
- Occasionally audio for a call ends up transmitting on one handset and receiving on the other (F0025841)
- A dangling RTP stream resulting from a network error may be re-used on a new call (F0031814)
- A missed call to an Avaya bridged or line appearance is not logged in the missed call register (F0031831)
Known Defects/Issues in Version 2.800.2.0
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SIP Interface Versions in Version 2.800.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.800.1.0
- Global muting with iCS telephone calls supported
- A1 & A2 LEDs alternatively flash green to indicate the device is in Soft Client mode
- DTMF tones supported in Soft Client mode
- DND configuration via the Soft Client interface supported
- Call forwarding configuration via the Soft Client interface supported
Defects Resolved in Version 2.800.1.0
- In soft client mode, the turret drops calls when changing to a speaker page where the appearance
for the call is associated with a speaker channel (F0031631)
- iD808 will go out-of-sync when it loses network connection while in soft client mode (F0031651)
- When in Soft Client mode audible incoming call ringing is not always heard (F0031661)
- With fast conferencing enabled and using Cisco Ad-hoc Conferencing, when making a standard 3-way
conference the conference ends up on hold and the handset is shown in the fast conferencing mode (F0031698)
Known Defects/Issues in Version 2.800.1.0
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SIP Interface Versions in Version 2.700.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.700.6.0
- Double-tap speaker functionality added
- Default handset as a Speaker source functionality added
- Missed calls shown on MWI made configurable via iCMS
- Font added to support Thai language demos
- Updated German translations
Defects Resolved in Version 2.700.6.0
- When an inbound call takes the device out of screen saver there is a few seconds delay
before the Caller ID is visible on the UI and if immediately answered a few seconds
delay before the audio is connected (F0030989)
- MTFIF resource leak when operating in Soft Client mode which can result in the iD808
losing registration with iCS (F0031377)
- Alerting issues with DMVS MRD Rx Ring (F0031397)
- In soft client mode, when answering a ringing call to an iCS call appearance the call
states stays in the ringing state for approximately a minute (F0031415,F0031484,F0031485)
- An MRD call doesn't go private if on a speaker channel and moved to a handset with
privacy pre-selected (F0031434)
- There is an audio burst out of the physical iTurret for a split second after the Soft
Client web client disconnects (F0031450)
- There is no acoustic shock protection in Quiet office mode (F0031478)
- Possible UI crash when in soft client mode if the messaging gateway contact IP address
is invalid when attempting to send a notify (F0031483)
- Changing the CTI Server network service hostname in iCMS does not change the hosts file entry (F0031537)
- Sidetone is not enabled when selecting MRD/Hoot DMVS appearances to assign them from
a speaker channel to a handset. This only occurs when no other users have the appearance
active on handsets or speaker channels and the speaker channel is latched on (F0031542)
- Possible iD808 crash in Soft Client mode (F0031581)
- When in soft client mode, call resources can be leaked resulting in the iD808 clearing
the soft client session and losing registration with iCS (F0031600)
- Sometimes, when the call on the handset is part of an active group and is put on hold
the other speaker channels which are members of the group do not get activated and stay
in the listening state (F0031618)
Known Defects/Issues in Version 2.700.6.0
- In soft client mode, the turret drops calls when changing to a speaker page where the appearance
for the call is associated with a speaker channel (F0031631)
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SIP Interface Versions in Version 2.700.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.700.5.0
- Updated Japanese translations
Defects Resolved in Version 2.700.5.0
- UI crash due to memory leak (F0030831)
- Voice activity may be shown on a soft client speaker when the call is not on the
speaker anymore (F0031312)
- When using soft client, clear down may fails after trying to retrieve a call (F0031313)
- When in private-elsewhere for a Cisco call the Bridged Call Appearance key label
incorrectly shows "Conference" (F0031349)
- Memory leak fix in MTFIF when dialling an iCS appearance and clearing it from
a handset (F0031353)
- When a CTI connection closes from the server side, the connection is not
automatically re-established (F0031383)
Known Defects/Issues in Version 2.700.5.0
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SIP Interface Versions in Version 2.700.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.700.4.0
- CTI interface supporting make, answer, clear, hold and unhold call
- Message waiting indicator LED enhanced to also illuminate for missed calls
(orange for missed calls, red for MWI)
Defects Resolved in Version 2.700.4.0
- When the device is starting up, the menu can be accessed before the screens have been drawn, resulting
in the black line overlay not being shown on screen A. A repower fixes the issue (F0030292)
- When in the Program Speaker Group configurator, if the back key is held down the user is returned
to key finder to select a group talk key instead of completely exiting the menu system (F0030296)
- Pressing a VPW on a speaker channel when there is an established call on a selected handset,
overrides the existing call and displays the VPW appearance. The original call can only be cleared
from the far end. Pressing the VPW appearance key works fine (F0030687)
- A turret configured with DMVS voice services may not always show the local/remote status in the
"Show SIP Server" status screen) (F0031002)
- Selecting a group talk key activates talk on a private elsewhere DMVS MRD call (F0031033)
- Group talk fails if the speaker source is set to handset 1 and the handset is busy with another
call (F0031034)
- Clearing a DMVS Hoot off the handset when part of an active group causes the hoot to get stuck
in the connecting state and doesn't transmit (F0031035)
- Speaker channels can show "attempting privacy" state when they should not (F0031077)
- When Local Muting is configured, after stop talk on any speaker channel other speaker channels
gets louder (F0031080)
- Cisco privacy doesn't work in soft client (F0031081)
- Occasionally, selecting a DMVS MRD appearance activates the call but the appearance key shows
busy elsewhere (F0031082)
- Sending a make call with a comma in the string to dial (the "dn" field) causes the turret to
send an "invalid message format received" which results in the call being stuck open at the
soft client end because the command rejected response then includes an "unknown" type in the
"rType" field. (F0031083)
- In soft client mode the labels displayed do not always match the turret. With handset 1 in
single line mode the short label was expected but the long label was displayed (F0031085)
- In soft client mode private busy elsewhere appearances can be selected but should be
rejected (F0031086)
- In soft client mode changing speaker page causes the microphone state to not match what happens
when not in soft client mode (F0031087)
- On a local endpoint, during login, with an MRD call already assigned to a speaker key
but not connected as this call is private elsewhere. At this point, if an endpoint at the
remote site is sending the ring signal the iD808 shows the "Action not possible" error
message (F0031088)
- DMVS MRD calls on speaker channels still send CDR events for incoming audio received even
when the call is private elsewhere (F0031113)
- When using LED scheme 2, DMVS MRD calls on speaker channel do not have the correct LED
colour when busy elsewhere (F0031117)
- Enabling privacy when a voice service is also connected somewhere doesn't show an error
in softclient (F0031126)
- In soft client mode, an attempt to assign a call on a handset to a read only speaker
channel is not rejected (F0031127)
- Can duplicate an assignment on the speaker channels when moved from handset (F0031140)
- When a turret starts up with a DMVS MRD on a speaker channel in the private elsewhere mode
and the user performs a group talk that includes that speaker channel, when the MRD changes
from private elsewhere to busy elsewhere the speaker channels automatically goes to the
talk state (F0031146)
- In soft client mode , when a handset is configured as single line mode and if there is a
directory entry for the answered call, it is displaying a Long label of the directory
entry rather than a short label (F0031147)
- In soft client mode, when a call is assigned from a handset to a empty speaker channel
and if there is already a speaker channel with the same appearance, the call is not assigned
to the empty speaker channel and is assigned to the existing speaker channel (F0031149)
- Pressing a speaker channel to answer a ringing VPW appearance or making an outbound call
with speaker source set to gooseneck causes the call to be moved to the selected handset
instead of staying on the speaker channel (F0031158)
- With an idle DMVS ARD call on a speaker channel and a call on the selected handset and
using gooseneck as the speaker source, pressing the speaker channel is rejected with
'Handset Busy' even though the call would have been made on the speaker channel (F0031159)
- Changing speaker page with a DMVS ARD in the busy elsewhere state causes the speaker
channels to become locked and cannot be selected and the second line of a dual line
display becomes blank (F0031169)
- When a speaker page contains a MRD DMVS appearance which is in private elsewhere state
and the speaker page is changed to a speaker page which doesn't contain this appearance,
the call entry is destroyed rather than being retained as a subscription only call.
This causes the display of duplicate call activity for the MRD DMVS call in soft
client mode (F0031177)
- When group talk button is pressed, the following states of calls on speaker channels
which are part of the group talk displays the green colour on the speaker channel LED
rather than the LED being off. 1) Busy elsewhere/Private elsewhere call state for
subscription only calls. 2) Private elsewhere call state for MRD DMVS (F0031208)
- Assigning a private avaya or cisco call to a speaker channel doesn't work in soft
client mode (F0031241)
- Turret crash after an attempted login to soft client (F0031258)
- In soft client, the call is cleared down when you move it from handset to speaker channel
on Mitel PBX (F0031261)
- In soft client, an occurrence of a call stuck on handset which cannot be cleared
(messaging protocol error) (F0031263)
- When the Soft Client receives a call from Elastix on an Elastix appearance you are
able to answer and hear two way audio but in Soft Client the call is still shown
as ringing (continuously) (F0031264)
- Using soft client, calls do not work properly after restarting the CUCM (F0031268)
- Soft client and Cisco CUCM v10.5, privacy causes handset bars to fail. Unable
to hangup handset. (CCD31269)
- Misleading message when trying to move a call that is on a handset onto a Speaker
Channel that is already occupied (F0031270)
Known Defects/Issues in Version 2.700.4.0
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SIP Interface Versions in Version 2.700.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.700.3.0
- Soft client session support extended to support two handsets, multiple speaker channels,
speed dials, voice services, hold and privacy functionality
- Global muting support added for Hoot voice services
- Dynamic keys enhanced to not add ringing calls if the alerting is disabled on the
associated appearance key
Defects Resolved in Version 2.700.3.0
- Occasional MTFIF crash (F0030736)
- The auto exit screen saver functionality doesn't work if a call is received just after another
call has stopped ringing (F0030989)
Known Defects/Issues in Version 2.700.3.0
- When in the Program Speaker Group configurator, if the back key is held down the user is
returned to key finder to select a group talk key instead of completely exiting the menu
system (F0030296)
- Pressing a VPW on a speaker channel when there is a call on the selected handset overrides
the existing handset call and displays the VPW appearance without clearing the original
call (F0030687)
- Clearing a connecting call off the handset in requesting privacy mode to move it to a speaker channel
does not remove the privacy request (F0031077)
- In soft client mode, when local muting is active, stopping talking on all channels correctly
unmutes the audio but the level is higher than when it was originally muted (F0031080)
- In soft client mode, privacy does not work on Cisco (F0031081)
- Occasionally, selecting a DMVS MRD appearance activates the call but the appearance key
shows busy elsewhere (F0031082)
- In soft client mode, sending a make call with a comma in the string to dial (the "dn" field)
causes the turret to send an "invalid message format received" which results in the call being
stuck open at the soft client end because the command rejected response then includes an "unknown"
type in the "rType" field. (F0031083)
- Taking a call off-hold on a speaker channel doesn't work in soft client mode (F0031084)
- In soft client mode the labels displayed do not always match the turret. (F0031085)
- In soft client mode private busy elsewhere appearances can be selected but should be rejected (F0031086)
- In soft client mode changing speaker page causes the microphone state to not match what
happens when not in soft client mode (F0031087)
- On a local endpoint, during login, if there is an MRD call already assigned to a speaker key
but not connecte as this call is private elsewhere. At this point, if an endpoint at the remote
site is sending the ring signal the iD808 shows the "Action not possible" error message (F0031088)
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SIP Interface Versions in Version 2.700.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.700.2.0
- No features or enhancements added to this release
Defects Resolved in Version 2.700.2.0
- With screen saver auto exit enabled, if an incoming call takes a unit out of screen saver and then
the call clears, if the user presses a key the unit still goes back into screen saver 10 seconds
after the call clears (F0030295)
- A listen only DMVS hoot which is set to not be recorded may still be mixed into the voice
recording stream (F0030298)
- Sometimes when creating a soft client session, the RTP stream used in place of the microphone
may not be connected resulting in one-way audio (F0030305)
Known Defects/Issues in Version 2.700.2.0
- When in the Program Speaker Group configurator, if the back key is held down the user is
returned to key finder to select a group talk key instead of completely exiting the menu
system (F0030296)
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SIP Interface Versions in Version 2.700.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.700.1.0
- Soft client session support (basic call functionality only)
- Echo cancellation for the two handsets
- Single key press to move an intercom handsfree call to a handset
- Removal of free key wizard
- Handset mute state updated after the handset mode is changed to push-to-talk or push-to-mute
- DTMF tone volume reduced by 6dBs
- User name displayed on an idle intercom splash screen
- Auto Discovery enhanced to blank out the user name when the user is logged out
Defects Resolved in Version 2.700.1.0
- The iCMS status message "Profile error - No valid master Avaya appearance found" and a yellow iCMS
icon may be incorrectly displayed after removing all Avaya appearances via a live update. A
resync clears the error (F0030048)
- When "Ring on busy" is configured as off and the user is in an intercom call using the handsfree
intercom interface, audible alerting is heard for ringing calls. The same applies if the user
is busy on the non-selected handset (F0030066)
- The Cisco PBX is not unregistered when the user logs out (F0030150)
- With "Ring on busy" is configured as off, alerting is silenced when a handset is busy but they are
not reinstated if the call on the handset is assigned or returned to a speaker channel. Similarly
a ringing alert is not silenced if a call on a speaker channel is assigned to a handset (F0030194)
- The Voice Mail menu item is greyed out unless a user has an Avaya call appearance configured (F0030243)
- DMVS MRDs or ARDs which are part of a speaker group will not work correctly when the group talk
key is selected (F0030261)
Known Defects/Issues in Version 2.700.1.0
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SIP Interface Versions in Version 2.600.9.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.9.0
- Screen saver auto-exit enhanced to stop entering the screen saver when there is a call on any of the
handsets or on the intercom handsfree interface
- Screen saver auto-exit enhanced to delay reactivating the screen saver by 10 seconds at the end
of an active call or an alerting call
- The Ethernet MTU value is now configurable between 1000 and 1500 bytes via iManager
Defects Resolved in Version 2.600.9.0
- When auto exit screen saver is enabled, receiving a ring on a DMVS MRD call for a remote
unit does not exit the screen saver. Local units work correctly (F0030104)
- For local users, IGMP leave messages are not sent for any multicast addresses used in
DMVS calls unless the multicast address was in use when the call was ended or the user
does not have the voice service on a key. This means the endpoint will continue to receive
all SbRTP for the active calls (but not play any audio) even if the appearance is not on
a handset or speaker channel (F0030105)
Known Defects/Issues in Version 2.600.9.0
- The iCMS status message "Profile error - No valid master Avaya appearance found" and a yellow iCMS
icon may be incorrectly displayed after removing all Avaya appearances via a live update. A
resync clears the error (F0030048)
- When "Ring on busy" is configured as off and the user is in an intercom call using the handsfree
intercom interface, audible alerting is heard for ringing calls (F0030066)
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SIP Interface Versions in Version 2.600.8.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.8.0
- No features or enhancements added to this release
Defects Resolved in Version 2.600.8.0
- When using an Asterisk PBX and the Voicemail Policy is set to none the message waiting
indicator is not displayed (F0014641)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- An Adhoc conferencing attempt on Avaya Aura may fail because the "To" and "From" tags
in the SIP REFER message are reversed (F0029838)
- On Avaya, when answering a ringing call on a bridged or line appearance, handset
mute may not work correctly (F0029899)
- For DMVS ARD calls when Ringback Tone Generation is configured as "Deskstation" the
iD808 does not generate the ringback tone (F0029902)
- "bMtfifTellSAUISubEventMultiValuesNotify() failed" error message generated when
unsubscribing to iCS during a log out or resync. Also there is a 500 internal server
error response to the Expires=0 NOTIFY from iCS. (F0029909)
- If a DMVS ARD call to the far end is made private and put on hold by a device and then
that device is resynchronised the call cannot be taken off-hold. It can only be cleared down
by the far end (F0029911)
- CVE-2015-0235 "GHOST" vulnerability in C library (F0029956)
- When a live update is received to disable global muting for an DMVS ARD on a speaker
channel in a call, the microphone does not work on the call until the call is ended
and re-created (F0029964)
- Live updates to talk/listen rights for DMVS Hoot appearances on speaker channels
or handsets do not take affect until the call is fully cleared or the device is
synchronised (F0029965)
- If a DMVS ARD appearance is deleted (via the UI or iCMS) in the busy elsewhere state,
the line goes idle and the appearance is added back in, then the key continues to show
the line as busy elsewhere but cannot be barged into (F0029966)
- When using iE801s and fast conferencing is enabled, adding an MRD or hoot first may
generate DSP errors and audio to subsequent conference members will not work (F0029967)
- Adding DMVS appearances to a conference when in fast conference mode may not work (F0029968)
- When DMVS Hoot or MRD is assigned to a turret (but not to the gateway), the SIP icon
is yellow, the DMVS Hoot or MRD appearance keys are shown as inactive, but nothing is
reported in SIP Server status to tell you the DMVS Hoot or MRD`s are the cause of
the yellow sip icon (F0029979)
- When a turret is attempting to connect to a hoot or MRD channel but this constantly
failing because the iCS resource limit has been reached, the turret is sluggish
to respond to anything (F0029982)
- Specific IGMP membership query messages are not handled resulting in multicast not
being received for up to 125 seconds (F0029984)
- When there is a loop on the network and the device receives the multicast it is
transmitting it will correctly attempt to change the SSRC and continue transmitting,
however this can result in voice services getting stuck busy elsewhere where streams
are not correctly removed (F0030004)
- UI errors are produced when L1 and L2 are in an ARD call and L1 attempts to put the
call on hold. The call is correctly cleared but errors are generated reporting that
the RTP call link could not be created (F0030007)
- When a manual ring is received on a DMVS MRD call which is in the busy elsewhere private
state it is incorrectly shown as ringing on dynamic keys and the appearance key allowing
the user to attempt to answer it (F0030008)
- There is a memory leak if an attempt to retrieve a Cisco configuration file from the
CUCM fails without the TFTP server reporting an error (F0030010)
- The "Failed to DNS continue on a call" internal error message is generated when
attempting to barge-in on a DMVS ARD call in the connecting state (F0030032)
- After a live update to remove an appearance an alert tone may continue to ring if the
appearance was ringing at the time it was removed (F0030033)
Known Defects/Issues in Version 2.600.8.0
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SIP Interface Versions in Version 2.600.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.7.0
- No features or enhancements added to this release
Defects Resolved in Version 2.600.7.0
- On an Avaya PBX, sometimes an attempt to convert a transfer into a conference fails (F0029818)
- When Dynamic Keys Auto-Refresh is disabled and a DMVS MRD call is active on the handset, receiving
the ring from the far end shows the call ringing on a dynamic key (F0029821)
Known Defects/Issues in Version 2.600.7.0
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
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SIP Interface Versions in Version 2.600.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.6.0
- No features or enhancements added to this release
Defects Resolved in Version 2.600.6.0
- A DMVS ARD doesn`t show as busy elsewhere on a local turret when another local turret
is in a calling out state (F0029735)
- Privacy doesn`t work with Avaya Aura registered appearances (F0029751)
- Barge-in doesn`t work with Avaya Aura registered appearances (F0029754)
- No ringing is heard on a DMVS MRD when it`s on a speaker channel; it works fine when it`s
not on the speaker channel. (F0029770)
- When barging-in to an ARD on a speaker channel which has not been answered at the
far end the barge in correctly fails but an alert tone is heard and the dynamic key
flashes with the busy-elsewhere call being removed and added several times (F0029773).
- When the handset is in push to answer mode, MRD calls are not answered (F0029780)
- Sometimes when answering a ringing DMVS MRD, the dynamic key continues to show
the ringing call for a few seconds even though the call is active on the handset (F0029784)
- A hoot voice service doesn`t change over to SbRTP from DMVS with SbRTP and perhaps
DMVS when changed via iCMS (F0029791)
Known Defects/Issues in Version 2.600.6.0
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
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SIP Interface Versions in Version 2.600.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.5.0
- Screen saver auto exit mode enhanced to have a configurable timer and for an alerting call not to
restart the screen saver timeout period
Defects Resolved in Version 2.600.5.0
- Registration may fail to an Asterisk PBX (F0027971)
- When a dynamic key with a DMVS ARD busy elsewhere call is barged in to, the busy status on
the dynamic key may still be shown (F0029336)
- When the network connection to the iG330 devices are lost, the iD808 DMVS appearances flash
between red and black for a period of time, then settle down and stop (can be up to 5 minutes) (F0029394)
- When registration is lost on iCS or iG330 network is disrupted, the handsfree/gooseneck
lamp on the turret flashes for a bit then stops (F0029395)
- Possible UI crash if the device is repowered when creating configuration files (F0029402)
- When 2 users attempt to select a DMVS voice service at the same time one will fail and the
user will need to select the appearance again. For SbRTP appearances selecting a voice
service always works. This could be improved by retrying on the endpoint (similar
to auto barge in on speaker) (F0029407)
- When an iCMS profile error is reported in the iCMS status screen the error is always
shown in English regardless of the display language (F0029411)
- The dynamic keys shows a shared appearance as still being on-hold after it has been taken off-hold on
another Turret with the shared appearance for a call made across a SIP trunk between Mitel
and iCS (F0029418)
- On a local device when barging into an MRD in the busy elsewhere state (shown on a dynamic key) by
pressing either the appearance key or dynamic key the call becomes active but the dynamic key
continues to show the call as busy elsewhere (F0029459)
- The dynamic key does not update when an MRD has been cleared off a speaker channel (F0029494)
- DMVS voice services may be incorrectly reported as out of service on the SIP status page (F0029525)
- When a DMVS ARD is on a speaker channel and the user stops talking the ARD call is incorrectly
displayed on a dynamic key (F0029526)
- An occurance of no transmit audio on a DMVS Hoot on a speaker channel on local turret (F0029543)
- Repeatedly selecting a speaker channel on a local endpoint with a DMVS call (with at least
2 other users active in the call) causes the speaker channel to get locked in the listening
mode and the user unable to talk by pressing the speaker channel (F0029552)
- When in a SIP point-to-point telephone call to iD808 devices running code before
2.600.3.5 with VAD enabled, the DSP on the device running earlier firmware will crash
when there is silence on the call (F0029557)
- If a call goes from alerting to busy-elsewhere it is possible for a dynamic key to
show the busy-elsewhere call using the alert colour style (F0029560)
- A live update of the labels of a DMVS voice service does not update any keys (F0029568)
- When a DMVS voice service is moved to a speaker for the first time, audio may not be
heard on the speaker channel (F0029589)
- When using DMVS appearances, global muting only works on speaker channels (F0029601)
- Live updates to active DMVS appearances to change global muting do not work until the
call is fully cleared (F0029613)
- When barging into a DMVS call on a local endpoint the microphone may not be activated.
On the handset the user may need to mute then unmute the handset to activate the microphone (F0029614)
- On remote turrets for DMVS ARD or MRD calls, global muting is incorrectly active when
listening on a speaker channel (F0029635)
- If a live update is received straight after a request to send the log files is processed
then the logs can get stuck in the enabled state and cannot be sent again until the
device is repowered (F0029637)
- Some SIP rejection messages are not immediately handled, causing a delay in any response action (F0029670)
- There are some major issues with conferencing with DMVS voice services (when they are
on speaker channels) (F0029706)
- When listening to a DMVS MRD on a speaker channel, if another user makes the call private
the VAD indicator still updates when the user starts or stops talking even though the audio
is muted (F0029720)
Known Defects/Issues in Version 2.600.5.0
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
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SIP Interface Versions in Version 2.600.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.4.0
- SIP status screen changed to remove region ID from iCS registration
- SIP status screen changed to list any out of service voice services and take the
SIP status icon yellow
- Timezone data updated for Moscow, Fiji, Istanbul, Tel Aviv and Damascus
Defects Resolved in Version 2.600.4.0
- When using DMVS appearances with VAD enabled global muting may not work correctly (F0029292)
- DMVS appearances on speaker channels are not displayed on or removed from dynamic keys when
they become active. Also, when they are shown on dynamic keys the style and indicator are
often incorrect. (F0029309)
- Cannot move a conference on a speaker to another speaker (F0029327)
- "No valid master appearance found" in the iCMS status screen may be misleading because
it does not specify if it is the Avaya master appearance or the iCS master appearance (F0029337)
- The MCC details are only shown on the SIP Server Status screen if an intercom appearance is
configured on the device. MCC is also used for the DMVS voice services (F0029338)
- A newly provisioned DMVS voice service may be shown as out of service until the device
is resynchronised (F0029350)
- Possible iD808 crash for local devices when configured with DMVS Hoot/MRD and no SIP call
appearance. Selecting the Hoot/MRD appearance crashes the turret when a SIP call is attempted (F0029351)
- Possible LCC crash when processing DSP alerts (F0029403)
- The dial tone does not move when you move the call from handset 1 to handset 2 (F0029410)
- An occurance of an audio feed not being heard on a local turret after doing a number of config
changes in iCMS (F0029415)
- For DMVS calls, if a media change to use SbRTP is received while the call is on a handset
then a slot in the DSP will be allocated but never removed resulting in the DSP appearing
to be full and new calls using SbRTP will not work. When in this state the device will need
to be repowered. Calls on speaker channels should be fine and this only affects calls on handsets (F0029422)
- An occurance of a hoot on a speaker channel that does not work (F0029424)
- An occurance of a hoot on a speaker channel that can receive but not transmit (F0029425)
- Memory leak when subscribing to non-iCS PBX (e.g. Avaya) due to new functionality added in version 2.6 (F0029432)
- When a DMVS hoot is on a speaker channel with the microphone not active, assigning the call
to the handset (via the assign key or with the speaker source set to selected handset) correctly
move the call but does not enable transmit (F0029433,F0029488)
- When changing speaker page the DSP can be configured incorrectly if one of the appearances
which is on a speaker channel is also on a handset (F0029452)
- An occurance of no audio heard in either direction on a DMVS hoot on a turret on a speaker channel (F0029454)
- iD808 incorrectly reports the MCC as down when the multicast pool is zero (F0029491)
- The dynamic keys do not update when an MRD is busied out on another turret (F0029493)
- When using an SbRTP ARD with VAD enabled on a speaker channel, unlatching the speaker channel
to deactivate the microphone incorrectly disables the transmit which clears down the call (F0029510)
Known Defects/Issues in Version 2.600.4.0
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- When a dynamic key with a DMVS ARD busy elsewhere call is barged in to, the busy status on
the dynamic key may still be shown (F0029336)
- When the network connection to the iG330 devices are lost, the iD808 DMVS appearances flash
between red and black for a period of time, then settle down and stop (can be up to 5 minutes) (F0029394)
- When registration is lost on iCS or iG330 network is disrupted, the handsfree/gooseneck
lamp on the turret flashes for a bit then stops (F0029395)
- Possible UI crash if the device is repowered when creating configuration files (F0029402)
- When 2 users attempt to select a DMVS voice service at the same time one will fail and the
user will need to select the appearance again. For SbRTP appearances selecting a voice
service always works. This could be improved by retrying on the endpoint (similar
to auto barge in on speaker) (F0029407)
- When an iCMS profile error is reported in the iCMS status screen the error is always
shown in English regardless of the display language (F0029411)
- The dynamic keys shows a shared appearance as still being on-hold after it has been taken off-hold on
another Turret with the shared appearance for a call made across a SIP trunk between Mitel
and iCS (F0029418)
- On a local device when barging into an MRD in the busy elsewhere state (shown on a dynamic key) by
pressing either the appearance key or dynamic key the call becomes active but the dynamic key
continues to show the call as busy elsewhere (F0029459)
- The dynamic key do not update when an MRD has been cleared off a speaker channel (F0029494)
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SIP Interface Versions in Version 2.600.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.3.0
- Quiet Office mode admin lock removed
- The upgrader ".gz" file may now be in either the upgraders folder or the root directory
of the TFTP server
Defects Resolved in Version 2.600.3.0
- With auto hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Occassional iD808 crash when adding Cisco call appearances (F0029132)
- Possible loss of incoming audio on a SbRTP voice service after a repower (F0029164)
- When using DMVS appearances, if SbRTP critical settings are changed such as bandwidth
type or packet size, DMVS calls are cleared down and MRD/Hoot appearances end up in
the inactive state and cannot be selected until the device is repowered or synchronised (F0029198)
- When a DMVS ARD private call is put on hold, other devices show the appearance as changing
from elsewhere private to on hold and no longer show the call as private. Barge in is
still rejected by iCS but the user is not showing the correct state (F0029219)
- When an active DMVS call is not in the established state (e.g. on hold or in a
conference) the call will clear after a few minutes (F0029220)
- If a call uses multicast media at any point the call info page will continue to show
the multicast address for the duration of the call, even if it is now using unicast (F0029230)
- When in fast conference mode (flashing handset LED), selecting an MRD or Hoot appearance
key to assign the call to the handset works fine but when the call is then assigned
to a speaker channel the handset continues to flash and receive and transmit audio
using the handset and not move to the speaker channel (F0029241)
- When using a listen only DMVS hoot, selecting the appearance key sends the microphone
input to the voice recorder until the microphone is muted (F0029276)
Known Defects/Issues in Version 2.600.3.0
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- When using DMVS appearances with VAD enabled global muting may not work correctly (F0029292)
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SIP Interface Versions in Version 2.600.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.2.0
- "Call Information" screen enhanced for Hoot and PWs
Defects Resolved in Version 2.600.2.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- The DSP has to be reset after an iE801 MAC address is programmed (F0027274)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- Selecting a speed dial on a remote turret configured to use an intercom appearance shows the
warning "Action not possible. Connecting to backup server. Please wait..." (F0028878)
- With a hoot, MRD and two ARDs (all SbRTP) assigned to speaker channels 1-4 and all four added to a
conference, when the conference is put on hold and taken off hold by selecting a speaker channel,
the call can only be assigned to the handset by selecting the speaker channel used to take the
conference off hold. Selecting other speaker channels shows "Action Not Possible" (F0028880)
- If DMVS calls are unavailable (because the gateway is not registered to iCS) the appearance
keys are not shown as "unavailable" (red icons) and the user only knows there is an issue when
they attempt to select the appearance and get an "Action not possible" message (F0028881)
- When the loud listen mode is changed, handsfree/loud listen is not disabled resulting in
the handset being shown in "loud listen" mode when loud listen is disabled (F0028882)
- With speaker source set to handset 1 and a hoot call on a speaker channel latched on, if handset
2 is selected and the appearance key is selected then the call moves to handset 2 but talk does
not work (F0028885)
- With quiet office enabled, pressing a speed dial to an intercom number (P2P or group) uses
the intercom splash screen and not the configured handset (F0028886)
- When assigning a DMVS MRD from the handset to a speaker channel which is displaying an
unavailable voice service (ARD without multicast configured) the speaker channel cannot be
selected to enable talk (F0028887)
- For local users, when iCS is restarted auto barge-in of MRD or Hoots will fail if the gateway
registers before the endpoints register. This is because the barge in is attempted before the
user has registered and it is rejected (F0028888)
- Initial status updates received for MRD or Hoot voice services for local user changes the appearance
key to busy elsewhere if it is displayed on the current page. The busy elsewhere appearance is not
shown on a dynamic key and changing page redraws the keys as idle (F0028889)
- When a call is in handsfree or loud listen mode and is cleared from the far end, handsfree or
loud listen are not disabled on the idle handset. When the user clears the call with either of
these enabled they are disabled (F0028890)
- When adding a new directory entry address in the personal directory the address above the one
entered is selected (F0028891)
- When the user edits an idle PBX appearance key that is also on a speaker channel to change the
style of the appearance, the appearance key is correctly updated but the speaker channel
continues to show the old style (F0028893)
- Changing an appearance key which is linked to a speaker channel key from dual mode to single line
mode also changes the speaker channel to single line mode (F0028894)
- The "Transmit Gain Offset" only works when the handset/headset is configured as a handset (F0028927)
- With the selected handset idle and the other handset configured as the default handset, there
are issues when dialling out from a speed dial (F0028953)
- When a DMVS MRD call is on a speaker channel and talk is not active, if another user makes
the call private the "private conversation" can still be heard on the speaker channels (F0029051)
- With two users barged in to a DMVS ARD call with global muting enabled and neither with
their microphone enabled, neither can hear the audio from the remote end (F0029004)
- When in fast conference mode (selected handset flashing green), when a Cisco hidden privacy
call is answered it is incorrectly added to the conference meaning fast conference cannot
be exited then entered again when the handset is idle. Attempting to enter fast conference
displays "Conference unavailable" message (F0029024)
- When a call is bridged on both handsets, clearing the call off the selected handset
doesn't update the soft keys (F0029025)
- For DMVS ARD or MRD calls, selecting an appearance key with privacy pre-selected on the
handset does not enable privacy and the call stays in the requesting private state.
Privacy needs to be removed then re-applied before the call is made private (F0029050)
- When paging up in the Call Info screen, after wrapping around to the bottom of the screen
the next up key press does not move the scroll position. Also when paging down, when the
bottom is reached if the up key is pressed followed by the down key, the next key press does
not move the scroll position (F0029070)
- Sometimes the UI can crash when receiving the manual ring for DMVS calls (F0029085)
- MRD ring received on speaker in dual line mode changes the 2nd line to "VS_GW_x" (F0029086)
- If a barge in attempt fails, the MTFIF call entry is not released resulting in the device
eventually running out of lines and being unable to make or receive calls. (F0029108)
- An orphaned DMVS call will clear after a few minutes and change the display to "nobody" (F0029109)
Known Defects/Issues in Version 2.600.2.0
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With auto hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
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SIP Interface Versions in Version 2.600.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.600.1.0
- DMVS support added
- Remote turret support for private wire and hoot channels
- Headset bypass relay control support added
- Remote syslog added
- Push-to-answer handset mode modified to ensure the handset is not muted at the start of a call
- Call register size increased to 40 entries
- "Missed Calls" shortcut to menu option added
- "Placed Calls" call register enhanced to only display the most recent call to each called number
- "Default Handset" configuration supported
- 12/24 hour clock configuration
- Click-to-dial via SIPTAPI demo supported
- SBRootAdmin password changed
Defects Resolved in Version 2.600.1.0
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- The SNMP service does not work when configured with an SNMP server with a 15 character long IP
address (e.g. 100.100.100.100) (F0027471)
- With dynamic key auto refresh enabled, incoming ARD or MRD calls may be displayed on two dynamic
keys; one showing the call as busy elsewhere and the other showing the call as ringing (F0027718)
- If a user presses a key at exactly the same time as the screen saver starts to activate it is possible
for the turret to get into a state where none of the cached keys are displayed (F0027869)
- The turret screen saver is de-activated when a call is made to an Avaya bridged appearance which
is not assigned to a key on the turret (F0027907)
- It is possible to add duplicate appearance keys via key finder (F0027908)
- The SNMP Mode, IP Address and Community String reported in the Network Setting strings report
the values that were in use when the device was last powered up, logged in or resynced but does
not reflect any changes that may have occurred due to a live update. (F0027935)
- The iD808 fails to register to an Avaya PBX when using a domain name as the registrar address (F0027936)
- Occasional UI crash when unregistering a PBX (F0028028)
- With "Allow DND" enabled in iManager, if the first menu the user enters after power-up is the
"Call preference" menu the DND option is shown in red. (F0028051)
- When pressing a nav key to move pages, if iCMS resynchronises the device it is possible for the auto key
repeat not to be cancelled with the result that the iD808 will continuously change pages until a key
is pressed to cancel the auto repeat timer. (F0028056)
- Occasional UI crash caused by calls to g_string_append_printf() and g_strdup_printf() with NULL
strings (F0028134)
- The iD808 does not successfully enable VAD when requested to by the PBX as the device checks for
"Silencesup" but most PBXs send out "silenceSup" resulting in the device assuming it is OFF (F0028159)
- Transfer fails when connected across an iCS trunk when using non-default port numbers (F0028462)
- Sometimes call forwarding is not enabled correctly on the turret (F0028569)
- Soft keys are not updated when dialling from the personal directory or call register (F0028873)
- When taking an intercom P2P call off hold to the handset the sidetone is not updated (F0028874)
- Short labels containing an '&' do not display correctly on speaker channel keys (F0028736)
- When the speaker source is set to handset 1 or handset 2 and a telephone appearance is linked to
a speaker channel and the speaker channel has tap latching enabled, taking a call off hold by tapping
the speaker channel puts the call on the handset but the call cannot be cleared by pressing the
handset clear key. The call needs to be "unlatched" by pressing the speaker channel key (F0028875)
- When a call on the handset is also assigned to a speaker if incoming audio stops while the call is
on the handset and then within 5 seconds incoming audio starts again and the call is cleared to the
speaker channel a CDR event for incoming audio stop is sent out even when audio is being
received (F0028801)
- Local muting is not fully displayed in the call preferences menu (F0028810)
- When the speaker source is set to handset 1 or handset 2, bridge handset functionality works when
it should be blocked (F0028811)
- When an on-hold call is taken off-hold by another user, the call is not shown as busy-elsewhere
on a dynamic key of the original user. (F0028876)
- When a private Cisco call on the selected handset is assigned to a speaker channel, privacy is removed
from the call but the idle handset remains in the requesting privacy state (F0028877)
- Live update to a dynamic key to remove busy-elsewhere call types leaves the busy-elsewhere call on the
dynamic key (F0028879)
Known Defects/Issues in Version 2.600.1.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With auto hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- Selecting a speed dial on a remote turret configured to use an intercom appearance shows the
warning "Action not possible. Connecting to backup server. Please wait..." (F0028878)
- With a hoot, MRD and two ARDs (all SbRTP) assigned to speaker channels 1-4 and all four added to a
conference, when the conference is put on hold and taken off hold by selecting a speaker channel,
the call can only be assigned to the handset by selecting the speaker channel used to take the
conference off hold. Selecting other speaker channels shows "Action Not Possible" (F0028880)
- If DMVS calls are unavailable (because the gateway is not registered to iCS) the appearance
keys are not shown as "unavailable" (red icons) and the user only knows there is an issue when
they attempt to select the appearance and get an "Action not possible" message (F0028881)
- When the loud listen mode is changed, handsfree/loud listen is not disabled resulting in
the handset being shown in "loud listen" mode when loud listen is disabled (F0028882)
- With speaker source set to handset 1 and a hoot call on a speaker channel latched on, if handset
2 is selected and the appearance key is selected then the call moves to handset 2 but talk does
not work (F0028885)
- With quiet office enabled, pressing a speed dial to an intercom number (P2P or group) uses
the intercom splash screen and not the configured handset (F0028886)
- When assigning a DMVS MRD from the handset to a speaker channel which is displaying an
unavailable voice service (ARD without multicast configured) the speaker channel cannot be
selected to enable talk (F0028887)
- For local users, when iCS is restarted auto barge-in of MRD or Hoots will fail if the gateway
registers before the endpoints register. This is because the barge in is attempted before the
user has registered and it is rejected (F0028888)
- Initial status updates received for MRD or Hoot voice services for local user changes the appearance
key to busy elsewhere if it is displayed on the current page. The busy elsewhere appearance is not
shown on a dynamic key and changing page redraws the keys as idle (F0028889)
- When a call is in handsfree or loud listen mode and is cleared from the far end, handsfree or
loud listen are not disabled on the idle handset. When the user clears the call with either of
these enabled they are disabled (F0028890)
- When adding a new directory entry address in the personal directory the address above the one
entered is selected (F0028891)
- When the user edits an idle PBX appearance key that is also on a speaker channel to change the
style of the appearance, the appearance key is correctly updated but the speaker channel
continues to show the old style (F0028893)
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SIP Interface Versions in Version 2.511.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.511.1.0
- No features or enhancements added to this release
Defects Resolved in Version 2.511.1.0
- On Avaya, when answering a ringing call on a bridged or line appearance, handset mute may not work correctly (F0029899)
Known Defects/Issues in Version 2.511.1.0
- Meet-Me conferencing does not work with Cisco Call Manager 8.6 (F0022659)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When initiating a Group Call from the iD808 the announcement tone heard at the iD712 may be
quieter than if the call was initiated from the iD712/SE708 (F0025910)
- When a call is in loud listen mode and is cleared from the far end the call failed tone is only
heard on the handset and not the main speaker (F0025986)
- When a call is in loud listen mode, DTMF tones are only played on the handset and not the
main speaker (F0025987)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- When changing the SIP port number on the turret and then doing a resync, the turret resyncs twice (F0026988)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- The SNMP service does not work when configured with an SNMP server with a 15 character long IP
address (e.g. 100.100.100.100) (F0027471)
- With dynamic key auto refresh enabled, incoming ARD or MRD calls may be displayed on two dynamic
keys; one showing the call as busy elsewhere and the other showing the call as ringing (F0027718)
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SIP Interface Versions in Version 2.510.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.51
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.510.1.0
- Quiet Office mode added for evaluation
Defects Resolved in Version 2.510.1.0
- When MCC is disabled the turret still attempts an IGMP join to the multicast address 239.60.19.2 (F0028655)
- When searching for a directory match, outbound dialling rules are not applied to the directory address string
meaning that matches are not found unless the directory is programmed with the same address which needs to
be sent to the PBX or is received from the PBX. (F0028781)
- When using iD808 with Cisco CUCM v10 call forwarding can be enabled but not disabled (F0028323)
Known Defects/Issues in Version 2.510.1.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- The SNMP service does not work when configured with an SNMP server with a 15 character long IP
address (e.g. 100.100.100.100) (F0027471)
- With dynamic key auto refresh enabled, incoming ARD or MRD calls may be displayed on two dynamic
keys; one showing the call as busy elsewhere and the other showing the call as ringing (F0027718)
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SIP Interface Versions in Version 2.501.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.501.1.0
- No features or enhancements added to this release
Defects Resolved in Version 2.501.1.0
- Occasional crash when creating an Avaya Adhoc conference (F0028090)
- No voice recording audio is sent on stream 1 if a device is set up with
no audio sources selected for any voice recording stream (F0027680, F0027829)
Known Defects/Issues in Version 2.501.1.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- The SNMP service does not work when configured with an SNMP server with a 15 character long IP
address (e.g. 100.100.100.100) (F0027471)
- With dynamic key auto refresh enabled, incoming ARD or MRD calls may be displayed on two dynamic
keys; one showing the call as busy elsewhere and the other showing the call as ringing (F0027718)
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SIP Interface Versions in Version 2.500.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.7.0
- No features or enhancements added to this release
Defects Resolved in Version 2.500.7.0
- Subscriptions may not be renew if a re-subscription attempt is rejected by the server (F0027724)
- When using auto-locate iCMS via DHCP with no port configured in the DHCP server the port number
is set to zero and therefore will fail to connect to iCMS (F0027771)
Known Defects/Issues in Version 2.500.7.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- The SNMP service does not work when configured with an SNMP server with a 15 character long IP
address (e.g. 100.100.100.100) (F0027471)
- No voice recording audio is sent on stream 1 if a device is set up with
no audio sources selected for any voice recording stream (F0027680)
- With dynamic key auto refresh enabled, incoming ARD or MRD calls may be displayed on two dynamic
keys; one showing the call as busy elsewhere and the other showing the call as ringing (F0027718)
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SIP Interface Versions in Version 2.500.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.6.0
- Improved icons for "Conference-hold private" and "Handset idle handsfree" in style 1 and
"Conference-hold" and "Seized-elsewhere requesting private" in style 2
Defects Resolved in Version 2.500.6.0
- No defects resolved in this release
Known Defects/Issues in Version 2.500.6.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
- The SNMP service does not work when configured with an SNMP server with a 15 character long IP
address (e.g. 100.100.100.100) (F0027471)
- No voice recording audio is sent on stream 1 if a device is set up with
no audio sources selected for any voice recording stream (F0027680)
- With dynamic key auto refresh enabled, incoming ARD or MRD calls may be displayed on two dynamic
keys; one showing the call as busy elsewhere and the other showing the call as ringing (F0027718)
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SIP Interface Versions in Version 2.500.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.5.0
- Enhanced icon for Dynamic Key (busy-elsewhere and held calls)
Defects Resolved in Version 2.500.5.0
- Infrequent UI lockup when making multiple changes in iManager including disabling fixed key or user
page editing (F0027441)
Known Defects/Issues in Version 2.500.5.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
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SIP Interface Versions in Version 2.500.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.4.0
- No features or enhancements added to this release
Defects Resolved in Version 2.500.4.0
- The UI can crash in the freekey wizard if at the same time the administrator edits a directory or key
in iManager (F0026276)
- When the ‘Allow Personal Directory Editing’ privilege setting is de-selected, it is still possible to
copy a call record from the call register to the personal directory. The "Copy" option should be
greyed out. (F0027334)
Known Defects/Issues in Version 2.500.4.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- Immediate Transfer to a mobile device from a natively registered Mitel appearance on an iD808
endpoint is unreliable (F0027356)
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SIP Interface Versions in Version 2.500.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.3.0
- No features or enhancements added to this release
Defects Resolved in Version 2.500.3.0
- Ring and mute and ring once does not work correctly when the incoming call is received via a
SIP NOTIFY as is the case for an Avaya bridged appearance (F0027245)
- When Dynamic Key Auto-Refresh is enabled, menu Auto Hide functionality does not work correctly (F0027261)
- When an update is received immediately after an invite to change call information (such
as the display name) the information may be ignored if the UI has not processed the
initial invite message (F0027262)
Known Defects/Issues in Version 2.500.3.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
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SIP Interface Versions in Version 2.500.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.2.0
- Support for new user privileges for "Allow DND", "Allow Fixed Key Editing", "Allow User Page Editing",
"Allow Alert Profile Editing" and "Allow Personal Directory Editing"
- "Screen Saver Auto-Exit" mode enhanced to action any key press that exits the screen saver
- Call register enhanced to display the line ID
- New alert modes of "Ring once" and "Ring and mute" added
- Default key style for anonymous appearance keys changed to dual line
Defects Resolved in Version 2.500.2.0
- When downgrading from v2.5 to an earlier version of iD808, the iD808 may fail to connect to iCMS
immediately after the downgrade (F0026710)
- The gooseneck active LED is not enabled when it should be for intercom calls using the intercom
splash screen when Loud Listen is enabled and Hands-Free Microphone type is set to Gooseneck or
Gooseneck Exclusive from the User Preferences menu (F0026756)
- Sometimes when starting up the MCC communication does not work (F0026760)
- There is no Music on Hold (MoH) generated when doing either a transfer or conference (F0026766)
- An occurance of a dangling call leg when a conference that is on-hold is cleared down (F0026767)
- An iD808 can go "out-off-sync" when reinstalling iCMS. The iD808 is logged out and when a user
is seated the iD808 will be back in sync (F0026779)
- When adding or editing a directory entry and in the type select screen, holding down the back key
only returns to the directory entry screen and not exit the menu (F0026794)
- Voice services that have the "Record" box not ticked in iCMS still generate CDR events with the
"should_record" flag set to TRUE when the microphone is active or audio is being received (F0026813)
- UI crash when attempting to switch between Cisco SRST and Cisco CUCM during failover (F0026987)
- With call pickup enabled and having answered a previous call on a dynamic key, any future missed calls
on the shared line appearance appear in the received calls list instead of the missed calls list
in the call register (F0027001)
- When auto select idle handset is enabled, selecting a call from the redial list when there is a call on
the selected handset in the far end ringing state does not work correctly (F0027059)
- When moving a connecting ARD call from one handset to the other the ringback tone is moved to the correct
handset but the speaker/microphone connection is not moved so the wrong handset continues to work until
the call is cleared (F0027131)
- Calls in the connecting state do not allow DTMF tones to be sent (F0027164)
- When answering shared lines on Avaya, the call ID supplied by the PBX is always used and no directory look
up or inbound dialling rules are applied to the number. This can cause other users with the appearance
shared to see something different to the user that answered the call (F0027165)
Known Defects/Issues in Version 2.500.2.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
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SIP Interface Versions in Version 2.500.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.500.1.0
- Slate UX user interface
- Locate iCMS using DHCP
- Japanese UI supported
- Intercom Audio Device supported
- Default styles for new keys (to match iManager)
- SIP port configuration
- Login changed to logon
Defects Resolved in Version 2.500.1.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When a Hoot or MRD is assigned to a speaker channel from iManager and not from the
turret no CDR events are sent until the appearance key is pressed to move the call
to the handset. Assigning the call from the speaker channel to the handset (via
assign+speaker channel) does not generate the CDR events. Synchronising,
re-powering or changing speaker page corrects the issue (F0026147)
- When the turret is waiting for a logout to fully configure a network change,
upgrading the firmware will fail. However the network configuration change
will be applied and subsequent upgrade attempts will be successful (F0026189)
- After a live update to add a speaker channel, clearing a conference call off
the handset to the new speaker channel clears the call but locks up the speaker
channel and appearance key (F0026211)
- Sometimes when starting up with iE801s attached the network icon can stay yellow
and the network status page report something like "iE801#1 DSP reported 2 link
restarts" however this does not appear to cause any issues (F0026240)
- Occasionally an iCS appearance key label still shows the Caller ID of the far end after
the call has been cleared (F0026241)
- The UI can crash in the freekey wizard if at the same time the administrator
edits speed dials in iManager (F0026276)
- The user prompt for group calls is not translated into the local language (F0026321)
- The intercom splash screen is not redrawn in the new locale when a live update
changes the locale (F0026329)
- The intercom splash screen is not cleared when the user is logged out if the
intercom line is in the line seized state (F0026331)
- IP addresses with all 1 digit sub-fileds (E.g. 1.2.3.4) will not be accepted
from iCMS and converted to 0.0.0.0 (F0026479)
- When starting up, CMSIF immediately starts communicating with iCMS which can cause issues
if iCMS sends an update to synchronise the device or change some settings (F0026672)
- Shared lines that are answered by other users are shown in the missed call
register and show the missed call icon (F0026673)
Known Defects/Issues in Version 2.500.1.0
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
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SIP Interface Versions in Version 2.200.7.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.7.0
- No features or enhancements added to this release
Defects Resolved in Version 2.200.7.0
- Occasionally an iCS appearance key label still shows the Caller ID of the far end after
the call has been cleared (F0026241)
Known Defects/Issues in Version 2.200.7.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- When a Hoot or MRD is assigned to a speaker channel from iManager and not from the
turret no CDR events are sent until the appearance key is pressed to move the call
to the handset. Assigning the call from the speaker channel to the handset (via
assign+speaker channel) does not generate the CDR events. Synchronising,
re-powering or changing speaker page corrects the issue (F0026147)
- When the turret is waiting for a logout to fully configure a network change,
upgrading the firmware will fail. However the network configuration change
will be applied and subsequent upgrade attempts will be successful (F0026189)
- After a live update to add a speaker channel, clearing a conference call off
the handset to the new speaker channel clears the call but locks up the speaker
channel and appearance key (F0026211)
- Sometimes when starting up with iE801s attached the network icon can stay yellow
and the network status page report something like "iE801#1 DSP reported 2 link
restarts" however this does not appear to cause any issues (F0026240)
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SIP Interface Versions in Version 2.200.6.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.6.0
- No features or enhancements added to this release
Defects Resolved in Version 2.200.6.0
- When Dynamic Keys Auto-Refresh is enabled, the dynamic keys may not update correctly when an active call
is cleared by the PBX to change to the busy elsewhere state (F0026556)
Known Defects/Issues in Version 2.200.6.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- When a Hoot or MRD is assigned to a speaker channel from iManager and not from the
turret no CDR events are sent until the appearance key is pressed to move the call
to the handset. Assigning the call from the speaker channel to the handset (via
assign+speaker channel) does not generate the CDR events. Synchronising,
re-powering or changing speaker page corrects the issue (F0026147)
- When the turret is waiting for a logout to fully configure a network change,
upgrading the firmware will fail. However the network configuration change
will be applied and subsequent upgrade attempts will be successful (F0026189)
- After a live update to add a speaker channel, clearing a conference call off
the handset to the new speaker channel clears the call but locks up the speaker
channel and appearance key (F0026211)
- Sometimes when starting up with iE801s attached the network icon can stay yellow
and the network status page report something like "iE801#1 DSP reported 2 link
restarts" however this does not appear to cause any issues (F0026240)
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SIP Interface Versions in Version 2.200.5.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.5.0
- "Use key style" configuration option for alert profiles
Defects Resolved in Version 2.200.5.0
- No defects resolved in this release
Known Defects/Issues in Version 2.200.5.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- When a Hoot or MRD is assigned to a speaker channel from iManager and not from the
turret no CDR events are sent until the appearance key is pressed to move the call
to the handset. Assigning the call from the speaker channel to the handset (via
assign+speaker channel) does not generate the CDR events. Synchronising,
re-powering or changing speaker page corrects the issue (F0026147)
- When the turret is waiting for a logout to fully configure a network change,
upgrading the firmware will fail. However the network configuration change
will be applied and subsequent upgrade attempts will be successful (F0026189)
- After a live update to add a speaker channel, clearing a conference call off
the handset to the new speaker channel clears the call but locks up the speaker
channel and appearance key (F0026211)
- Sometimes when starting up with iE801s attached the network icon can stay yellow
and the network status page report something like "iE801#1 DSP reported 2 link
restarts" however this does not appear to cause any issues (F0026240)
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SIP Interface Versions in Version 2.200.4.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.4.0
- No features or enhancements added to this release
Defects Resolved in Version 2.200.4.0
- With "Dynamic Keys Auto-Refresh" enabled, when ringing calls are first added to dynamic keys they
are added to the first available empty dynamic key and do not take into account the alert
priority of the appearance key (F0026354)
Known Defects/Issues in Version 2.200.4.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- When a Hoot or MRD is assigned to a speaker channel from iManager and not from the
turret no CDR events are sent until the appearance key is pressed to move the call
to the handset. Assigning the call from the speaker channel to the handset (via
assign+speaker channel) does not generate the CDR events. Synchronising,
re-powering or changing speaker page corrects the issue (F0026147)
- When the turret is waiting for a logout to fully configure a network change,
upgrading the firmware will fail. However the network configuration change
will be applied and subsequent upgrade attempts will be successful (F0026189)
- After a live update to add a speaker channel, clearing a conference call off
the handset to the new speaker channel clears the call but locks up the speaker
channel and appearance key (F0026211)
- Sometimes when starting up with iE801s attached the network icon can stay yellow
and the network status page report something like "iE801#1 DSP reported 2 link
restarts" however this does not appear to cause any issues (F0026240)
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SIP Interface Versions in Version 2.200.3.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.3.0
- No features or enhancements added to this release
Defects Resolved in Version 2.200.3.0
- When the Loud Listen function is activated the gooseneck mic is disabled (as
required) but the level LEDs still light up as normal if you speak near the
mic (F0026229)
Known Defects/Issues in Version 2.200.3.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
- When a Hoot or MRD is assigned to a speaker channel from iManager and not from the
turret no CDR events are sent until the appearance key is pressed to move the call
to the handset. Assigning the call from the speaker channel to the handset (via
assign+speaker channel) does not generate the CDR events. Synchronising,
re-powering or changing speaker page corrects the issue (F0026147)
- When the turret is waiting for a logout to fully configure a network change,
upgrading the firmware will fail. However the network configuration change
will be applied and subsequent upgrade attempts will be successful (F0026189)
- After a live update to add a speaker channel, clearing a conference call off
the handset to the new speaker channel clears the call but locks up the speaker
channel and appearance key (F0026211)
- Sometimes when starting up with iE801s attached the network icon can stay yellow
and the network status page report something like "iE801#1 DSP reported 2 link
restarts" however this does not appear to cause any issues (F0026240)
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SIP Interface Versions in Version 2.200.2.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.2.0
- U-Boot htest 30 enhanced to display pictograms for key presses
- Improved German translations
Defects Resolved in Version 2.200.2.0
- Low audio level may be heard during an Avaya Adhoc conference (F0026027)
- Cannot cancel privacy after bridging a call between handsets (F0026028)
- RTP streams are left active after an Avaya Adhoc conference (F0026031)
- iD808 crashes when using two line appearances and attempting to transfer one of the calls (F0026049)
- When deleting a "Float key" or a "Group Talk key" and returning from the confirm
action help screen the text on the confirm screen lists "unknown" instead of the key type (F0026094)
- When a loud listen function key in iManager is sent to a turret that does not support
loud listen the key is shown as a DND function key but does not function correctly (F0026104)
- When receiving an ARD call it can appear on two dynamic keys (F0026107)
Known Defects/Issues in Version 2.200.2.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
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SIP Interface Versions in Version 2.200.1.0
- Avaya Interface Version 2.20
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.200.1.0
- Avaya Ad-hoc conferencing
- Transfer to conference (with Avaya Ad-hoc conferencing)
- Call pickup from line (with Avaya)
- Dynamic key automatic refresh
- "Transfer only" outbound dialling restrictions option
- Tone-to-line speed dial type
- Push-to-answer
- Loud listen
- Screen saver exit on incoming call
- German language support
Defects Resolved in Version 2.200.1.0
- When attempting to transfer a call, resources may be lost in the SIP stack if the original call
is cleared before the transfer is completed (F0025347)
- Voice recording calls can start and not stop when using MRD calls on speaker channels (F0025560)
- After a live update of a Cisco appearance, incoming calls to that appearance may fail to ring (F0025701)
- When attempting to answer a call on an Avaya bridge appearance by pressing the appearance key, if
the appearance is configured to not allow outbound calls the key press is rejected with an
"Outbound calls restricted" warning message (F0025770)
- Functions, page and menu shortcut keys are not automatically refreshed when the locale is
changed with a live update (F0025789)
- There is no validity checking for intercom dial numbers when creating or editing personal directory
entries (F0025814)
- When adding calls to a conference via the fast conference functionality, it is possible to add
private calls (F0025874)
- When configured with backup Cisco PBXs and an SRST server that is not available the iD808 can
get stuck attempting to register to SRST and never attempting to register to the primary
PBX again (F0025883)
- Sometimes calls do not get added to dynamic keys when the state is changed e.g. busy-elsewhere
to on-hold-elsewhere (F0025886)
- If the iD808 has two registrations (i.e. Avaya for PSTN and iCS for intercom) but does not have
the intercom appearances assigned to keys, calls for both PSTN and intercom will ring the Avaya call
appearance if the numbers match (F0025889)
- When adding new Lines from the iD808 (when intercom appearances are configured but not assigned)
the add list is incorrectly formatted (F0025890)
- Speed dials starting with the '/' character do not use client side conference but are added
to an Ad Hoc Conference if supported (F0025974)
Known Defects/Issues in Version 2.200.1.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
- The SIP stack may fail to initialise if there are more than 5 DNS servers configured (F0025297)
- When several users have the same shared appearance on an auto answer speaker channel
only one of the users is guaranteed to auto answer the call (F0025432)
- Unattended transfer calls that have been missed may have the incorrect number listed
in the call register (F0025544)
- When the SIP logs are enabled the device can run out of memory if left on permenantly (F0025675)
- When the speaker source is set to handset 1 or handset 2 (or selected handset and a group
talk key has been pressed) and there are multiple calls on the handset, enabling handsfree
does not work correctly for all calls. There is also a similar issue when enabling loud listen (F0025988)
- When in push to answer mode, if a ringing call on a shared line is attempted to be
answered on both handsets at the same time then a warning message may be displayed but
the call will be answered on one handset successfully. If there are two ringing calls
then pressing both handset switches at the same time to attempt to answer both calls
will only answer the first ringing call (F0025989)
- With Auto Hold enabled and auto select idle handset enabled, repeatedly pressing
the redial key can get the device stuck with calls in the far end ringing state but
not on either handset and cannot be cleared by the user and need to be answered by
the far end to be cleared down (F0025990)
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SIP Interface Versions in Version 2.111.4.0
- Avaya Interface Version 1.40
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.111.4.0
- No features or enhancements added to this release
Defects Resolved in Version 2.111.4.0
- Occasional failure of the SIP stack to initialise due to a DNS interface initialisation error (F0025297)
- Occasional one-way voice on an Avaya PBX (F0025324)
- Ringback not heard during an unattended transfer with a Cisco PBX (F0025329)
Known Defects/Issues in Version 2.111.4.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
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SIP Interface Versions in Version 2.111.3.0
- Avaya Interface Version 1.40
- Cisco Interface Version 2.10
- Mitel Interface Version 2.10
New Features/Enhancements Added in Version 2.111.3.0
- No features or enhancements added to this release
Defects Resolved in Version 2.111.3.0
- When transferring a call on a Cisco PBX, the call register stores the address
of the second leg with the label from the first leg (F0025141)
- When attempting to transfer a call on a Cisco PBX to a busy number the
second leg of the call only partially clears down leaving a blank label on
the handset and the first leg on hold. The user then has to press the
"Clear" key to fully clear down the second leg and return the first leg to
the handset (F0025153)
- With a Mitel PBX, L1 calls L2, L2 answers, L1 attended transfer to L3, both L2
and L3 shows Caller ID of L1 (F0025209)
Known Defects/Issues in Version 2.111.3.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
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SIP Interface Versions in Version 2.111.2.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.111.2.0
- No features or enhancements added to this release
Defects Resolved in Version 2.111.2.0
- In some circumstances, when a call is originated from a handset which is
muted, it is possible for the user to be heard by the far end (F0025108)
Known Defects/Issues in Version 2.111.2.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
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SIP Interface Versions in Version 2.111.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.111.1.0
- No features or enhancements added to this release
Defects Resolved in Version 2.111.1.0
- The UI may crash when making an intercom call from the handsfree interface
if the intercom appearance key is attached to a speaker channel (F0025123)
Known Defects/Issues in Version 2.111.1.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
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SIP Interface Versions in Version 2.110.3.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.110.3.0
- Improved alert styles for UI scheme 1
Defects Resolved in Version 2.110.3.0
- The iD808 may stop subscribing if the SIP stack silently terminates the subscription request (F0024945)
- Alert preview does not show the correct background for the icon area (F0024946)
- When a call attempt fails because the far end is busy, the call is not logged in the placed call
directory (F0024947)
- The iD808 may crash when the "CM defaultall" command is executed (F0024961)
- Cannot barge in to a Cisco shared line if the "To" and the "From" addresses are the same (F0024962)
Known Defects/Issues in Version 2.110.3.0
- Sometimes with an unattended transfer to a mobile is actioned via an iCS trunk
line, the mobile rings when dialled and then stops when the call is handed off
from the iD808 to the mobile. Then a short period later the mobile rings once
again which shows completion of the transfer for a call between the PSTN and
the mobile (F0023829)
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SIP Interface Versions in Version 2.110.2.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.110.2.0
- Additional dynamic key types for "busy-elsewhere and ringing calls" and "busy-elsewhere,
on-hold and ringing calls"
- Device ID displayed on log in screen
- The user is returned to the intercom splash screen when exiting the menu
- Timestamp added to file name of transmitted log files
- Copy of messages files generated when enabling additional logging
- Retain any core files during a power cycle
- Infrastructure added for German language support
- Infrastructure added for UI schemes
Defects Resolved in Version 2.110.2.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
- With "Move to idle handset" set to "Bridged Handset" when making a call on the second
handset with the media sourced from an iE801 module and then bridging the first
handset in and then clearing handset 1, the microphone audio of the headset is
heard on the speaker of the headset (F0024288)
- When an iD808 page is configured as read-only (or on a key page policy) it is not
possible to program the soft group talk key on the iD808 (F0024289)
- An appearance key stays in the busy elsewhere state if a user barges into a shared
line that is connected to a call appearance on the user's turret (F0024302)
- When an intercom appearance is the bottom key on screen A, dialling a P2P call and
clearing down, does not clear the call label and this label is then used to update
all call appearances on the turret (F0024380)
- When searching the directory hitting a letter and then using the down arrow causes
the prompt to jump back (F0024444)
- Very occasionally an ARD or MRD call can remain active for up to 5 seconds longer
than it should do (F0024508)
- An occurance of an iD808 crash when connected to two different iManagers (F0024561)
- Inter digit dialling does not work for intercom calls made from the intercom splash
screen or speaker channel. It only works for calls made on the selected handset (F0024580)
- When in the outbound dialling screen, entering a digit then pressing and releasing
the hash key enters the hash but after a second the hash is deleted and the key pad
incorrectly enters "ABC" mode (F0024581)
- When intercom privacy is enabled and an incoming call is displayed on the splash
screen, pressing the # key hides the splash screen and shows the call as on hold
but the far end is still connecting and the call continues to ring on the dynamic
key (F0024589)
- "Multiple iCS servers" error may be reported by the iD808 after increasing the
max appearances for an iCS server (F0024599)
- When pressing * and then seizing a line, dialpad digits still work as menu shortcuts
and do not start outbound dialling (F0024705)
- When rejoining an answerback group call it is not possible to answer the answerback
call by pressing * when prompted to do so (F0024706)
- Outgoing answerback groups calls on the handset that have the message dialog box
displayed do not clear down the message dialog box when the call is cleared by
the iCS server (by upgrade or restart) (F0024708)
- ARDs do not change from on-hold-elsewhere-private to on-hold-elsewhere when
privacy is deselected (by an SE708) (F0024744)
- When adding or inserting a new appearance key from key finder the line ID list
can be incorrectly filled in with extra blank lines when the user has not added
both intercom appearances to keys and the user does not have an intercom dial
number configured in iManager (F0024774)
- When attempting to transfer a call received on a shared line (on a bridged
appearance or line appearance) on Avaya 6.2 the far end does not receive music-on-hold
however if the transfer is aborted and re-made then music-on-hold then works fine (F0024848)
- When a user is seated on a device with no expansion modules fitted and no speaker
channels configured the Program Options for Speakers still allows the user to select
"Delete" "Edit" and "View" even though no speaker channels can be selected (F0024849)
Known Defects/Issues in Version 2.110.2.0
- No known defect or issues
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SIP Interface Versions in Version 2.110.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.110.1.0
- Hash to dial supported
- Inter-digit timer dialling supported
- Support added for a wider range of time zones including Asia/Kolkata
Defects Resolved in Version 2.110.1.0
- After pressing the conf key the user has to wait until the warning box clears down before being
able to dial (F0024445)
Known Defects/Issues in Version 2.110.1.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
- With "Move to idle handset" set to "Bridged Handset" when making a call on the second
handset with the media sourced from an iE801 module and then bridging the first
handset in and then clearing handset 1, the microphone audio of the headset is
heard on the speaker of the headset (F0024288)
- When an iD808 page is configured as read-only (or on a key page policy) it is not
possible to program the soft group talk key on the iD808 (F0024289)
- An appearance key stays in the busy elsewhere state if a user barges into a shared
line that is connected to a call appearance on the user's turret (F0024302)
- When an intercom appearance is the bottom key on screen A, dialling a P2P call and
clearing down, does not clear the call label and this label is then used to update
all call appearances on the turret (F0024380)
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SIP Interface Versions in Version 2.101.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.101.1.0
- No features or enhancements added to this release
Defects Resolved in Version 2.101.1.0
- When a bridged handset call is cleared from the far end it does not clear down correctly
and then there are issues with the next call made that can leave the appearance key stuck
in the busy state (F0024376)
Known Defects/Issues in Version 2.101.1.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
- With "Move to idle handset" set to "Bridged Handset" when making a call on the second
handset with the media sourced from an iE801 module and then bridging the first
handset in and then clearing handset 1, the microphone audio of the headset is
heard on the speaker of the headset (F0024288)
- When an iD808 page is configured as read-only (or on a key page policy) it is not
possible to program the soft group talk key on the iD808 (F0024289)
- An appearance key stays in the busy elsewhere state if a user barges into a shared
line that is connected to a call appearance on the user's turret (F0024302)
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SIP Interface Versions in Version 2.100.10.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.10.0
- No features or enhancements added to this release
Defects Resolved in Version 2.100.10.0
- The user interface on the iD808 may freeze after a directory entry, linked to a currently
displayed speed dial key, is deleted via iManager (F0024187)
- If the administrator attempts to upgrade an iD808 using an incorrect file which can be
found on a TFTP server the device gets stuck displaying the "upgrading your unit
please wait" screen until it is repowered (F0024260)
Known Defects/Issues in Version 2.100.10.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.9.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.9.0
- No features or enhancements added to this release
Defects Resolved in Version 2.100.9.0
- Sometimes when dropping out of an iCS conference call on a speaker channel
the speaker channel key returns to idle instead of busy-elsewhere. The
associated appearance key correctly shows busy-elsewhere (F0023602)
- Speed dials may be lost during a live update (F0024001)
- There may be a re-drawing issues when changing speed dials (F0024009)
- When in outbound dialling mode for the second leg of a transfer using Avaya
with auto-select-idle-handset enabled, leg 1 can be taken off hold which
selects the other handset and automatically hides the outbound dialling
screen. If both handsets are then cleared the menu cannot be accessed until
the device is repowered (F0024020)
- Local muting does not update correctly when an intercom call on the splash
screen is put on hold (F0024083)
- When selecting an appearance key for a shared line being taken off hold
by the user an error message can be added to the messages log file and
the call cleared (F0024092)
- A firmware upgrade fails without any indication that there is a problem if
the associated tar.gz file cannot be found on the TFTP server (F0024113)
- Cannot make outbound calls when registered directly to a Mitel PBX (F0024116)
Known Defects/Issues in Version 2.100.9.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.8.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.8.0
- Fast conferencing option improved to simultaneously support standard conferencing
Defects Resolved in Version 2.100.8.0
- When configured with Avaya call-forwarding-on-busy enabled, the device can end up
unable to make or receive calls if left long enough if the PBX responds with a 403
Forbidden message (F0023787 & F0023791)
- When conferencing calls, audio may not work for some members of a conference when
an intercom call is on hold (F0023922)
- Rejoining a group call might fail if there are a large number of group calls active (F0023927)
- Incoming calls on a bridge appearance may not show on a dynamic key for one of the appearances
on the iD808 (F0023959)
- On hold elsewhere calls cannot be added to conferences when in fast conference mode (F0023961)
- Fast conference mode becomes unavailable if the user is in fast conference mode and
the only call in the conference is cleared by the far end (F0023962)
Known Defects/Issues in Version 2.100.8.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.7.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.7.0
- No features or enhancements added to this release
Defects Resolved in Version 2.100.7.0
- Very occassionally when clearing out of an iCS barged in call on a speaker channel the
speaker key goes to the idle state even though the appearance key is still in the busy
elsewhere state (F0023602)
- When in the middle of an attended transfer if the user that is on-hold ends their call
followed by the user being called ending or answering their call, the call is still shown
as active on the UI for the originator of the call and has to be cleared manually (F0023691)
- No ringback tone is heard for the next call after a call has been transferred using an iCS
server (F0023700)
- If a line is seized then moved to the other handset, pressing a dialpad digit does not enter the
outbound dialling mode but is ignored. The seized line needs to be cleared then re-selected to
enter outbound dialling (F0023746)
- The conferencing mode option does not stay greyed out when some menu items are changed (F0023768)
Known Defects/Issues in Version 2.100.7.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.6.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.6.0
- The default Voice Recording Warning Tone level changed to 12dB lower than the original fixed level
Defects Resolved in Version 2.100.6.0
- Privacy may not work on Avaya (F0023719)
Known Defects/Issues in Version 2.100.6.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
- When in the middle of an attended transfer if the user that is on-hold ends their call
followed by the user being called ending or answering their call, the call is still shown
as active on the UI for the originator of the call and has to be cleared manually (F0023691)
- Very occassionally when clearing out of an iCS barged in call on a speaker channel the
speaker key goes to the idle state even though the appearance key is still in the busy
elsewhere state (F0023602)
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SIP Interface Versions in Version 2.100.5.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.5.0
- The Voice Recording Warning Tone level is now controlled by CM variables and the default level
is 6dB lower than the previous fixed level
Defects Resolved in Version 2.100.5.0
- Very occassionally when clearing out of an iCS barged in call on a speaker channel the
speaker key goes to the idle state even though the appearance key is still in the busy
elsewhere state (F0023602)
- After a live update to add an appearance key which is in the busy elsewhere state,
barging in to this line does not display "Conference" (F0023610)
- CDR issue where intermittently a heartbeat for a call with bridged handsets has missing
information (F0023612)
- The CDR recording details are not updated on the iD808 when the recorder setting is
changed to none in the relevant iTurret Recording Policy (F0023612)
- With auto clear enabled outgoing calls on the selected handset in the far end ringing
state are not cleared by pressing another appearance key. The far end has to answer
the call before auto clear works correctly. (F0023633)
- iD808 crash while selecting privacy on handset 2 with Avaya mis-configured (F0023654)
- Transfers fail when early media occurs on the second leg (F0023659)
- Asterisk VPWs do not work (F0023660)
Known Defects/Issues in Version 2.100.5.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
- When in the middle of an attended transfer if the user that is on-hold ends their call
followed by the user being called ending or answering their call, the call is still shown
as active on the UI for the originator of the call and has to be cleared manually (F0023691)
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SIP Interface Versions in Version 2.100.4.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.4.0
- No features or enhancements added to this release
Defects Resolved in Version 2.100.4.0
- If an intercom call is received on a speaker channel and terminates too quickly the turret will play
a constant busy tone (F0023506)
- Fast Conferencing does not work with Avaya 5.2 bridged appearances (F0023513)
- Cannot bridge handset with a client-side conference (F0023560)
Known Defects/Issues in Version 2.100.4.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.3.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.3.0
- Handset bridging
- Fast conferencing
Defects Resolved in Version 2.100.3.0
- No defects resolved in this release
Known Defects/Issues in Version 2.100.3.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.2.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.2.0
- Support for faster failover when using Cisco or Avaya PBXs
Defects Resolved in Version 2.100.2.0
- When a user is configured with intercom appearance but is seated on a site without an iCS,
the intercom appearance is correctly shows with the unavailable icon (red background)
however if the user attempts to view or edit the key via key finder then the Line ID
displayed is incorrect (F0023048)
- If an iD808 is upgraded before iCMS and then a live update is received and the iD808
is repower before a resync or a live update to a the channel XML file the iD808 can
crash during power up (F0023129)
- If an answerback group call is put on-hold and then another user answers back, the turret
can no longer make point-to-point intercom calls until it is re-powered (F0023171)
- The intercom appearance label is not shown after assigning a call from the speaker
channel to the handset (F0023172)
- Once a forced group call is assigned to a speaker channel their is no way to clear/wipe
it from the speaker channel (F0023173)
- An iD808 lock up can occur if a received group call is put on hold then a higher
priority group call is received (F0023181)
- Local muting does not update correctly when assigning intercom calls between the splash
screen and the handset and does not update correctly when clearing a seized intercom
line by pressing the i key (F0023183)
- If the call attempt for the second leg of a transfer fails, incorrect information is
sent when the iD808 attempts to take the first call leg off hold (F0023235)
Known Defects/Issues in Version 2.100.2.0
- When repowering the iD808 the missed calls indicator icon is shown if there are old
entries in the missed call register (F0023155)
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SIP Interface Versions in Version 2.100.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.100.1.0
- Mitel PBX supported for basic telephony
- Configuration of ringback tone for ARD calls
- iCMS team key pages support added
Defects Resolved in Version 2.100.1.0
- Misleading error message seen when attempting to use two Cisco Call Managers on the same
iD808. Only a single Cisco Call Manager is supported (F0022623)
- With an iCS server, quickly repeatedly seizing a line on a bridged appearance and clearing
can result in the bridged appearance permanately reporting busy elsewhere (F0022645)
- When the speaker source is set to handset 1 or handset 2 and a speaker channel is selected
to answer a ringing SIP call, handset muting may not work correctly if other speaker channels
are active (F0022707)
- When in the directory search screen the search position does not update when the dialpad key
is held down to change the letter to a number (F0022709)
- When adding a paginating key in key finder, the user can change pages but cannot change
to read-only pages (F0022710)
- When receiving a call on a bridged Avaya line from iCS the call label is shown in quotes (F0022711)
- Tapping a listen only speaker channel still updates the internal "latched" setting for
the speaker channel which stops the speaker source menu item from being available to edit (F0022712)
- Transfer may fail when registered to an Avaya PBX and the call is received via a SIP trunk
line from a Cisco PBX (F0022852)
- When a user is listening to an ARD on a speaker channel with the microphone muted and privacy
is pre-selected on the handset (grey padlock) when the user assigns the call to the handset
the call briefly goes private before clearing privacy. Also, if the user has requested privacy
when other local users are active but then puts the call on hold, the privacy request is not
cleared resulting in the call coming off hold when the final local user drops out (F0022872)
- When the handsfree microphone type is set to gooseneck and handsfree is pre-selected on
an idle handset, assigning a listen only hoot call from a speaker channel to the handset
still shows the gooseneck indicator as active even though the microphone is muted (F0022873)
- Live updates of voice service call settings (such as call labels or multicast address) does
not work for voice services on float keys. Synchronising fixes the issue (F0022930)
- When moving a lot of keys containing voice services from iManager it is possible to end
up with some of the keys disappearing. Synchronising will return the keys but result in
the voice services still not working correctly. Re-powering will set up the device
correctly again (F0024062)
Known Defects/Issues in Version 2.100.1.0
- When a user is configured with intercom appearance but is seated on a site without an iCS,
the intercom appearance is correctly shows with the unavailable icon (red background)
however if the user attempts to view or edit the key via key finder then the Line ID
displayed is incorrect (F0023048)
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SIP Interface Versions in Version 2.001.2.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.001.2.0
- No features or enhancements added to this release
Defects Resolved in Version 2.001.2.0
- When moving a lot of keys containing voice services from iManager it is possible to end
up with some of the keys disappearing. Synchronising will return the keys but result in
the voice services still not working correctly. Re-powering will set up the device
correctly again (F0024062)
Known Defects/Issues in Version 2.001.2.0
- Misleading error message seen when attempting to use two Cisco Call Managers on the same
iD808. Only a single Cisco Call Manager is supported (F0022623)
- With an iCS server, quickly repeatedly seizing a line on a bridged appearance and clearing
can result in the bridged appearance permanately reporting busy elsewhere (F0022645)
- When the speaker source is set to handset 1 or handset 2 and a speaker channel is selected
to answer a ringing SIP call, handset muting may not work correctly if other speaker channels
are active (F0022707)
- When in the directory search screen the search position does not update when the dialpad key
is held down to change the letter to a number (F0022709)
- When adding a paginating key in key finder, the user can change pages but cannot change
to read-only pages (F0022710)
- When receiving a call on a bridged Avaya line from iCS the call label is shown in quotes (F0022711)
- Tapping a listen only speaker channel still updates the internal "latched" setting for
the speaker channel which stops the speaker source menu item from being available to edit (F0022712)
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SIP Interface Versions in Version 2.001.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.001.1.0
- No features or enhancements added to this release
Defects Resolved in Version 2.001.1.0
- Incoming calls on a bridge appearance may not show on a dynamic key for one of the appearances
on the iD808 (F0023959)
- Speed dials may be lost during a live update (F0024001)
- There may be a re-drawing issues when changing speed dials (F0024009)
Known Defects/Issues in Version 2.001.1.0
- Misleading error message seen when attempting to use two Cisco Call Managers on the same
iD808. Only a single Cisco Call Manager is supported (F0022623)
- With an iCS server, quickly repeatedly seizing a line on a bridged appearance and clearing
can result in the bridged appearance permanately reporting busy elsewhere (F0022645)
- When the speaker source is set to handset 1 or handset 2 and a speaker channel is selected
to answer a ringing SIP call, handset muting may not work correctly if other speaker channels
are active (F0022707)
- When in the directory search screen the search position does not update when the dialpad key
is held down to change the letter to a number (F0022709)
- When adding a paginating key in key finder, the user can change pages but cannot change
to read-only pages (F0022710)
- When receiving a call on a bridged Avaya line from iCS the call label is shown in quotes (F0022711)
- Tapping a listen only speaker channel still updates the internal "latched" setting for
the speaker channel which stops the speaker source menu item from being available to edit (F0022712)
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SIP Interface Versions in Version 2.000.4.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.000.4.0
- No features or enhancements added to this release
Defects Resolved in Version 2.000.4.0
- With an iCS server, user A answers a line shared with B and C, user B barges in and A, B & C
all display "Conference", user C barges in and the "Conference" label on C changes to a
dial number (F0022585)
- When using iCS sometimes the SIP icon momentarily goes yellow with a subscription failure (F0022668)
- There is a possible memory leak when processing a notify message (F0022708)
Known Defects/Issues in Version 2.000.4.0
- Misleading error message seen when attempting to use two Cisco Call Managers on the same
iD808. Only a single Cisco Call Manager is supported (F0022623)
- With an iCS server, quickly repeatedly seizing a line on a bridged appearance and clearing
can result in the bridged appearance permanately reporting busy elsewhere (F0022645)
- When the speaker source is set to handset 1 or handset 2 and a speaker channel is selected
to answer a ringing SIP call, handset muting may not work correctly if other speaker channels
are active (F0022707)
- When in the directory search screen the search position does not update when the dialpad key
is held down to change the letter to a number (F0022709)
- When adding a paginating key in key finder, the user can change pages but cannot change
to read-only pages (F0022710)
- When receiving a call on a bridged Avaya line from iCS the call label is shown in quotes (F0022711)
- Tapping a listen only speaker channel still updates the internal "latched" setting for
the speaker channel which stops the speaker source menu item from being available to edit (F0022712)
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SIP Interface Versions in Version 2.000.3.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.000.3.0
- No features or enhancements added to this release
Defects Resolved in Version 2.000.3.0
- With the handsfree microphone type set to gooseneck, the gooseneck indicator is not lit when the
handsfree key is pressed when the handset is idle (F0022544)
- When manually answering an intercom call on the handset or speaker channel with intercom privacy
enabled the call information popup is displayed when it shouldn't be. Also if the intercom
appearance key is configured with alerting enabled then the alerting does not stop when the
call is answered (F0022545)
- If the user has received a mixing group call on the intercom splash screen with intercom latching
enabled the user can latch the microphone active by pressing the * key but if the user then holds
down the handsfree speaker key the microphone is not disabled until the handsfree speaker key is
released. The microphone should be disabled after the timeout has expired so that the user knows
it is safe to release the key without clearing the call (F0022546)
- When taking an intercom call off hold on a speaker channel with latching enabled and the latching
mode set to tap latch the microphone is not latched on. It is when taking SIP telephone calls off
hold on speaker channels (F0022547)
- When the user is logged out and the logs are requested from iCMS the log file is generated but it
does not contain the DSP status files (F0022548)
- A live update of outbound dialling rules on iCS doesn't work and requires the device to be
synchronised for the change to take affect. These settings can be changed for other PBXs without
the need of a synchronise (F0022549)
- With the maximum of 50 appearance keys for an iCS dial number and activity on all the keys, active,
ringing or busy elsewhere, the NOTIFY messages are rejected by the iD808 because they are larger
than the maximum size accepted by the SIP stack. Apart from the NOTIFY being discarded the iCS
continues to repeat sending the message and after reaching the maximum retry attempts limit will
unsubscribe the endpoint. (F0022555)
- When the handset is muted, the microphone is not muted when taking an intercom call off hold even
though the icon shows the microphone as muted (F0022583)
- When changing a personal directory entry type from a non-intercom type to an intercom type any
speed dials configured to use that directory entry do not fully update which stops them from
working until the device is synchronised (F0022584)
Known Defects/Issues in Version 2.000.3.0
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SIP Interface Versions in Version 2.000.2.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.000.2.0
- No features or enhancements added to this release
Defects Resolved in Version 2.000.2.0
- When an iD808 only has an intercom appearance set-up, the SIP Server icon is displayed in yellow
and the SIP Server Status reports a subscription error (F0022362)
- An incoming call is incorrectly rejected when DND is on (F0022393)
- With an idle intercom appearance on a speaker channel and speaker microphone type set to selected handset
and the menu displayed, when an intercom call is received up and down no longer navigate the menu but change
the page. The only way to remove the menu is to press Ok, followed by the back key (F0022405)
- When a VPW has been configured for a user but is not on a key and all other appearances have been added
to keys the user still has the option to add or insert a line but the outbound ID list is blank or
shows wrong information (F0022406)
- Ringing calls on dynamic keys are shown with the wrong animated bell and key style (F0022407)
- With a single intercom appearance and with point-to-point intercom calls having priority over group
calls, the user is incorrectly returned to a group call while the point-to-point is active (F0022419)
- Inbound and outbound dialling rules are applied to intercom lines when configured on an iCS PBX.
The rules should only be applied to telephone numbers (F0022468)
- When two devices have a shared line as the default appearance and both attempt to dial at the same
time one device will fail to seize the line with the warning "Call line seized has timed out" (F0022469)
- Various minor CDR issues (F0022470)
- Tap latching doesn't work for the first time after an active call is assigned to an empty speaker
channel by tapping an empty speaker channel to move it from the handset (F0022471)
- If the iE801 f1 key is used to bring the iD808 out of screen saver mode and configured as Wipe or
Clear then the menu is shown to unassign or clear a speaker channel (F0022472)
- When the iE801 f1 keys are configured as Wipe or Clear they can still be selected to show the clear
speaker menu even when the Wipe speaker and Clear speaker menu options are greyed out because there
are no speakers to clear (F0022473)
- With handsfree pre-selected on the handset and the handsfree microphone type set to gooseneck and
a speaker channel latched on, assigning the speaker channel to the handset shows the "multiple calls
on gooseneck" warning message even though only one call is active. When the handsfree microphone
type is set to gooseneck exclusive the user is blocked from assigning the call with the warning
"gooseneck busy" (F0022501)
- When iCS lines have users barged in and display "Conference" the user can still attempt to make the
call private which sometimes shows the grey padlock and a warning and other times is successful
and shows the yellow padlock. The iD808 should block the attempt to make the call private as
privacy should not be possible in a conference. (F0022502)
- Intercom calls received from the SB534 do not work (F0022504)
- If a SIP message containing SDP is received with a blank far end IP address MTFIF will crash (F0022505)
- With intercom privacy enabled and an intercom appearance on a speaker channel, audio may not work
when the user answers the call as fixed settings are used (F0022508)
- If audio is received at the same time as a call is cleared by the far end for a call on a speaker
channel then a CDR incoming audio start event can be generated after the CDR ended event has been
sent. This has been seen with a P2P intercom call from the SB534 (F0022518)
- For an intercom P2P call on the selected handset pressing privacy toggles the grey padlock for the
call. The privacy request should be blocked as it is not supported. This has been seen with a
intercom P2P call from the SB534 (F0022519)
Known Defects/Issues in Version 2.000.2.0
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SIP Interface Versions in Version 2.000.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 2.000.1.0
- iCS PBX supported for enhanced telephony. Supported features:
- Common lamping
- Barge-in
- Privacy
- Line seize
- Pickup of on-hold-elsewhere
- Bulk registration
- Multiple call, bridged & line appearances
- Up to 50 appearance keys per extension number
- Speed dials and VPWs
- Call register
- Attended and unattended transfer
- Call forwarding (locally while the user is logged in to the turret)
- Early media
- G.711 A-law & u-law coding
- Single SIP trunk
- Title bar moved up by three pixels
Defects Resolved in Version 2.000.1.0
- With an Avaya PBX, subscription failures may be seen after all appearances are deleted and
re-added. A resync resolves the issue (F0022188)
- After a live update to a voice service on a fixed key the call may not clear correctly when
clearing from a handset (F0022189)
- When a user dials a number and while still in the calling state rapidly moves the call to
the other handset, when the call is answered there may be no audio (F0022131)
- With Avaya, if a call is transferred by pressing the transfer key followed by a speed dial
key very quickly the transfer fails (F0022214)
- When entering text via the UI, such as editing a page title or a personal directory entry,
when pressing a dial pad key, followed by the back key, followed by the same dial pad key
again causes existing characters to be deleted and the character displayed to be the next
one in the list (F0022215)
- When editing a general page title, speaker page title or alert profile title if a new page
or profile is selected before the title is saved the user is prompted if they wish to save
the changes. If yes is selected and the same page or profile is viewed the title is shown
as blank in the config edit screen (F0022222)
- When editing a speaker page title if a new page is selected before the title is saved the
user is prompted if they wish to save the changes. If the back key is pressed at this point
a blank general menu is displayed which can only be exited by holding down the Exit key
for the quick menu exit period (F0022223)
- When making an answerback group call on the intercom splash screen with the handsfree microphone
set to gooseneck, the gooseneck microphone indicator is not disabled when * is pressed to
initiate the answerback. The handsfree indicator correctly changes from green to red but
the gooseneck indicator stays illuminated (F0022233)
- When using an intercom appearance on a speaker channel that is not set to latching to
initiate an answerback to owner group call and the speaker microphone source is set to
gooseneck once the group is ready for the initiator to speak the microphone is not open (F0022234)
- If a device is out of sync in iCMS when it starts up and iCMS immediately synchronises it,
then the UI can lock up causing the device to be stuck in the initialising network screen (F0022275)
- When clearing a conference off a speaker channel it is possible for the UI to crash (F0022276)
- When the device is logged in and intercom appearances are configured, removing the
network cable changes the SIP server status icon to yellow and not red. The network
and iCMS status icons correctly show red. If the device is left for long enough the
SIP server status icon will change to red (F0022277)
Known Defects/Issues in Version 2.000.1.0
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SIP Interface Versions in Version 1.411.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.411.1.0
- No features or enhancements added to this release
Defects Resolved in Version 1.411.1.0
- If a SIP message containing SDP is received with a blank far end IP address MTFIF will crash (F0022505)
Known Defects/Issues in Version 1.411.1.0
- With an Avaya PBX, subscription failures may be seen after all appearances are deleted and
re-added. A resync resolves the issue (F0022188)
- After a live update to a voice service on a fixed key the call may not clear correctly when
clearing from a handset (F0022189)
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SIP Interface Versions in Version 1.410.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.410.1.0
- Answerback group calls supported
- Talk only permissions for intercom group calls supported
- Talk only permissions for hoot calls supported
Defects Resolved in Version 1.410.1.0
- When a received group call is cleared down by the user at about the same time as it is
cleared down by the originator error messages are logged to the messages file (F0021678)
- With handsfree pre-selected an intercom call can end up on the handset in handsfree mode and
not on the intercom splash screen when pressing redial twice. Handsfree should be disabled
in this mode and the intercom call should end up on the handset (F0021707)
- When a received group call is on the intercom splash screen and a live update is received to
move the intercom appearance key the call can still be put on hold but is not shown on an
appearance key (because it is orphaned) and not shown on a dynamic key. Hold should be
blocked for orphaned calls (F0021708)
- When an intercom call is ophaned due to moving the appearance key via a live update and
then assigned to a speaker channel the appearance key is shown as attached to the speaker
channel when it is not (F0021711)
- When there is an active ophaned call resynchonising the device does not clear down the call (F0021712)
- With tap-latching enabled, when a VPW is answered on a speaker channel talk is not latched on (F0021749)
- After seizing an intercom appearance on an appearance key or a speaker channel and then
dialling out an intercom speed dial, the seized intercom appearance is not used for the speed
dial call (F0021762)
- Error message "Unassigned keys currently xx when logging out" generated while logging out (F0021781)
- Occassionally the iD808 does not switch back to a group call in the background (F0021782)
- With the handsfree intercom configured as gooseneck and a call on the handset, muting the
handset leaves the gooseneck indicators active (F0022032)
- Various CDR issues including missing incoming audio stop event when taking a call off hold,
missing audio start event when answering an ARD, missing incoming audio stop events when
clearing a conference and no recording information in events generated when a group call
is cleared (F0022036)
- With two intercom appearance keys not attached to speaker channels and a point-to-point
active when an incoming group call is received it rings but when the point-to-point is
moved to a handset the group call is not auto-answered. The same happens if a group call
is active and a point-to-point is received (F0022185)
- When the voice recording warning tone is enabled DTMF tones do not work (F0022187)
- When sendlogs has completed the menu is not redrawn to show the available menu items (F0022190)
- The grammer for the key finder text is incorrect (F0022191)
- The call information for orphaned calls shows a "Called Line" title but no data (F0022192)
Known Defects/Issues in Version 1.410.1.0
- With an Avaya PBX, subscription failures may be seen after all appearances are deleted and
re-added. A resync resolves the issue (F0022188)
- After a live update to a voice service on a fixed key the call may not clear correctly when
clearing from a handset (F0022189)
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SIP Interface Versions in Version 1.400.12.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.12.0
- No features or enhancements added to this release
Defects Resolved in Version 1.400.12.0
- When an inbound intercom call from the SB534 system is answered the gooseneck microphone indicator
lights up even when the handsfree microphone is configured for internal (F0022006)
- Immediate Transfer on Avaya Aura v6.2 is unreliably (F0022010)
Known Defects/Issues in Version 1.400.12.0
- When a received group call is cleared down by the user at about the same time as it is
cleared down by the originator error messages are logged to the messages file (F0021678)
- With handsfree pre-selected an intercom call can end up on the handset in handsfree mode and
not on the intercom splash screen when pressing redial twice. Handsfree should be disabled
in this mode and the intercom call should end up on the handset (F0021707)
- When a received group call is on the intercom splash screen and a live update is received to
move the intercom appearance key the call can still be put on hold but is not shown on an
appearance key (because it is orphaned) and not shown on a dynamic key. Hold should be
blocked for orphaned calls (F0021708)
- When an intercom call is ophaned due to moving the appearance key via a live update and
then assigned to a speaker channel the appearance key is shown as attached to the speaker
channel when it is not (F0021711)
- When there is an active ophaned call resynchonising the device does not clear down the call (F0021712)
- With tap-latching enabled, when a VPW is answered on a speaker channel talk is not latched on (F0021749)
- After seizing an intercom appearance on an appearance key or a speaker channel and then
dialling out an intercom speed dial, the seized intercom appearance is not used for the speed
dial call (F0021762)
- Error message "Unassigned keys currently xx when logging out" generated while logging out (F0021781)
- Occassionally the iD808 does not switch back to a group call in the background (F0021782)
- With two intercom appearance keys not attached to speaker channels and a point-to-point
active when an incoming group call is received it rings but when the point-to-point is
moved to a handset the group call is not auto-answered. The same happens if a group call
is active and a point-to-point is received (F0022185)
- When the voice recording warning tone is enabled DTMF tones do not work (F0022187)
- With an Avaya PBX, subscription failures may be seen after all appearances are deleted and
re-added. A resync resolves the issue (F0022188)
- After a live update to a voice service on a fixed key the call may not clear correctly when
clearing from a handset (F0022189)
- When sendlogs has completed the menu is not redrawn to show the available menu items (F0022190)
- The grammer for the key finder text is incorrect (F0022191)
- The call information for orphaned calls shows a "Called Line" title but no data (F0022192)
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SIP Interface Versions in Version 1.400.11.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.11.0
- No features or enhancements added to this release
Defects Resolved in Version 1.400.11.0
- When the handsfree microphone setting is set to gooseneck and an intercom P2P call is
received from an SB534 user onto the intercom splash screen the gooseneck is active but the
gooseneck indicator is not on (F0021754)
- Using Cisco CM8, sometimes cannot barge into a call if another turret uses has barged in
to the call and then put the call on hold (F0021766)
- When rejoining a standard group call with gooseneck as the handsfree microphone the
gooseneck indicator is illuminated even though talk is not available (F0021797)
- With auto clear enabled and a locked group call on the selected handset, if an incoming
telephony call is answered both calls end up on the selected handset (F0021799)
- A mixing group call that is listen only for a user still prompts the user to press * to talk (F0021800)
- With the handsfree mic configured as gooseneck and a mixing group call that is listen
only on the handset, if this call is moved to handsfree the gooseneck indicator is illuminated (F0021801)
- When a mixing group call that is listen only for a user is put on hold and then taken off
hold, the user has an active microphone (F0021802)
Known Defects/Issues in Version 1.400.11.0
- When a received group call is cleared down by the user at about the same time as it is
cleared down by the originator error messages are logged to the messages file (F0021678)
- With handsfree pre-selected an intercom call can end up on the handset in handsfree mode and
not on the intercom splash screen when pressing redial twice. Handsfree should be disabled
in this mode and the intercom call should end up on the handset (F0021707)
- When a received group call is on the intercom splash screen and a live update is received to
move the intercom appearance key the call can still be put on hold but is not shown on an
appearance key (because it is orphaned) and not shown on a dynamic key. Hold should be
blocked for orphaned calls (F0021708)
- When an intercom call is ophaned due to moving the appearance key via a live update and
then assigned to a speaker channel the appearance key is shown as attached to the speaker
channel when it is not (F0021711)
- When there is an active ophaned call resynchonising the device does not clear down the call (F0021712)
- With tap-latching enabled, when a VPW is answered on a speaker channel talk is not latched on (F0021749)
- After seizing an intercom appearance on an appearance key or a speaker channel and then
dialling out an intercom speed dial, the seized intercom appearance is not used for the speed
dial call (F0021762)
- Error message "Unassigned keys currently xx when logging out" generated while logging out (F0021781)
- Occassionally the iD808 does not switch back to a group call in the background (F0021782)
- With two intercom appearance keys not attached to speaker channels and a point-to-point
active when an incoming group call is received it rings but when the point-to-point is
moved to a handset the group call is not auto-answered. The same happens if a group call
is active and a point-to-point is received (F0022185)
- When the voice recording warning tone is enabled DTMF tones do not work (F0022187)
- With an Avaya PBX, subscription failures may be seen after all appearnces are deleted and
re-added. A resync resolves the issue (F0022188)
- After a live update to a voice service on a fixed key the call may not clear correctly when
clearing from a handset (F0022189)
- When sendlogs has completed the menu is not redrawn to show the available menu items (F0022190)
- The grammer for the key finder text is incorrect (F0022191)
- The call information for orphaned calls shows a "Called Line" title but no data (F0022192)
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SIP Interface Versions in Version 1.400.10.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.10.0
- Enhancement to support both CDR protocol versions 6.0 and 7.0
Defects Resolved in Version 1.400.10.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- CDR events have incorrect call information for incoming group calls (F0021806)
Known Defects/Issues in Version 1.400.10.0
- When a received group call is cleared down by the user at about the same time as it is
cleared down by the originator error messages are logged to the messages file (F0021678)
- With handsfree pre-selected an intercom call can end up on the handset in handsfree mode and
not on the intercom splash screen when pressing redial twice. Handsfree should be disabled
in this mode and the intercom call should end up on the handset (F0021707)
- When a received group call is on the intercom splash screen and a live update is received to
move the intercom appearance key the call can still be put on hold but is not shown on an
appearance key (because it is orphaned) and not shown on a dynamic key. Hold should be
blocked for orphaned calls (F0021708)
- When an intercom call is ophaned due to moving the appearance key via a live update and
then assigned to a speaker channel the appearance key is shown as attached to the speaker
channel when it is not (F0021711)
- When there is an active ophaned call resynchonising the device does not clear down the call (F0021712)
- With tap-latching enabled, when a VPW is answered on a speaker channel talk is not latched on (F0021749)
- When the handsfree microphone setting is set to gooseneck and an intercom P2P call is
received from an SB534 user onto the intercom splash screen the gooseneck is active but the
gooseneck indicator is not on (F0021754)
- After seizing an intercom appearance on an appearance key or a speaker channel and then
dialling out an intercom speed dial, the seized intercom appearance is not used for the speed
dial call (F0021762)
- Error message "Unassigned keys currently xx when logging out" generated while logging out (F0021781)
- Occassionally the iD808 does not switch back to a group call in the background (F0021782)
- When rejoining a standard group call with gooseneck as the handsfree microphone the
gooseneck indicator is illuminated even though talk is not available (F0021797)
- With two intercom appearance keys not attached to speaker channels and a point-to-point
active when an incoming group call is received it rings but when the point-to-point is
moved to a handset the group call is not auto-answered. The same happens if a group call
is active and a point-to-point is received (F0022185)
- With auto clear enabled and a locked group call on the selected handset, if an incoming
telephony call is answered both calls end up on the selected handset (F0021799)
- A mixing group call that is listen only for a user still prompts the user to press * to talk (F0021800)
- With the handsfree mic configured as gooseneck and a mixing group call that is listen
only on the handset, if this call is moved to handsfree the gooseneck indicator is illuminated (F0021801)
- When a mixing group call that is listen only for a user is put on hold and then taken off
hold, the user has an active microphone (F0021802)
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SIP Interface Versions in Version 1.400.9.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.9.0
- No features or enhancements added to this release
Defects Resolved in Version 1.400.9.0
- Locked group calls are received at a low volume (F0021675)
- The user can talk back on a listen only group call (F0021676)
- If a user is receiving a group call, then receives a higher priority group call
and clears it quickly they may not be returned to the lower priority group call (F0021704)
- If a large number (+200) of group calls are made or received, voice services may not work
the next time the device is synchronised (F0021705)
- Occasionally when making an intercom group call on the intercom splash screen the call gets
stuck with the connecting call icon (lightning strike) and the microphone is muted and cannot
be unmuted (F0021706)
- The group call announce duration is a lot longer for iD808s than it is for iD712s (F0021723)
Known Defects/Issues in Version 1.400.9.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- When a received group call is cleared down by the user at about the same time as it is
cleared down by the originator error messages are logged to the messages file (F0021678)
- With handsfree pre-selected an intercom call can end up on the handset in handsfree mode and
not on the intercom splash screen when pressing redial twice. Handsfree should be disabled
in this mode and the intercom call should end up on the handset (F0021707)
- When a received group call is on the intercom splash screen and a live update is received to
move the intercom appearance key the call can still be put on hold but is not shown on an
appearance key (because it is orphaned) and not shown on a dynamic key. Hold should be
blocked for orphaned calls (F0021708)
- When an intercom call is ophaned due to moving the appearance key via a live update and
then assigned to a speaker channel the appearance key is shown as attached to the speaker
channel when it is not (F0021711)
- When there is an active ophaned call resynchonising the device does not clear down the call (F0021712)
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SIP Interface Versions in Version 1.400.8.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.8.0
- No features or enhancements added to this release
Defects Resolved in Version 1.400.8.0
- When registered to SRST and A calls B and B transfer to C, A still shows the remote Caller ID
as B (F0021566)
- The originator of a client-side conference may not be able to speak to or hear the conference
after adding a hoot channel to the conference (F0021577)
- When receiving an auto answered point-to-point intercom call that is quickly cleared by the
originating end it is possible for the intercom call to still be shown active on the UI after
the call has ended (F0021587)
- Possible UI lockup when running automated intercom call testing (F0021588)
- Miscellaneous error logs seen in the messages log file (F0021589)
- Auto answer on speaker channels doesn't work when the intercom splash screen is displayed (F0021600)
- The indicator next to intercom appearance keys for received intercom group calls is off but it
is green for point to point calls and other active calls (F0021601)
- When a connecting intercom call is moved from the handset to the intercom handsfree interface
the ring back tone stops (F0021602)
- When an intercom group call is received when an intercom point to point call is active causing
the point to point call to be put on hold this causes problems if the point to point call is
already on hold or ringing (F0021603)
- After registering to a backup Avaya server, when the iD808 is resynchronised it first attempts
to register to the backup server instead of the primary server. (F0021629)
- The UI can crash during start up when attempting to add an iCS PBX (F0021630)
- P2P intercom calls cleared down from the SB534 are not fully removed from the device (F0021631)
- When dialling out a point-to-point intercom call that is rejected by iCS the ID808 fails to
free up the MTFIF line used for the call. When all 600 MTFIF lines are used up no more SIP calls
will be possible (FF0021632)
- Possible lockup during power up with a Cisco configuration if the iCMS attempts to send a
resync or a live update during the initialisation period (F0021646)
- Sometimes when receiving intercom P2P and intercom group calls the device can get into a state
where received group calls get stuck in the connecting state and audio is not received (F0021648)
- When conferencing calls together, the conference mixers in the DSP are sometimes not cleared
down when the conference is cleared which can result in unpredictable behaviour such as the
speaker not being connected to the conference or the links between the DSPs (if iE801s are used)
not being set up for other conferences (F0021649)
- Sometimes after an upgrade a DSP reset message is seen in the Network status screen and the
network icon goes yellow. The device appears to recover and work normally. (F0021651)
- When running an autoscript the device can crash when failing to answer an intercom call
on the splash screen. The crash occurs when attempting to add the call to the call
register (F0021666)
- When making intercom group calls with iE801s connected the audio might not work (F0021668)
Known Defects/Issues in Version 1.400.8.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
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SIP Interface Versions in Version 1.400.7.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.7.0
- No features or enhancements added to this release
Defects Resolved in Version 1.400.7.0
- With the Cisco CUCM offline and the iD808 registered to a Cisco SRST if the iD808 is
resynchronised the Cisco configuration is lost and the iD808 cannot register to the SRST (F0021528)
- There is a memory leak when checking a notify message for a directory match (F0021543)
- When SbRTP calls are added to a conference the Call Information screen does not show any information
for the call if the Circuit Reference has not been set for the call in iCMS (F0021553)
- When failed over to a Cisco SRST server and seizing a SRST appearance with the menu active the next
SRST appearance key displays an incoming call (F0021557)
Known Defects/Issues in Version 1.400.7.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
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SIP Interface Versions in Version 1.400.6.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.6.0
Defects Resolved in Version 1.400.6.0
- The labels from intercom directory entries may be used for telephony calls (F0021434)
- In a conference, for SIP calls, the call information in Device Info > Call Information
in the UI only shows the source number not the destination numbers (F0021465)
- Occassionally the originator of a client-side conference send out silence for one or
more call legs (F0021466)
- When receiving several simultaneous inbound group calls if an inbound locked group call
results in an intercom point-to-point call being put on hold there is a possibility that
the UI may lock up (F0021475)
Known Defects/Issues in Version 1.400.6.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- With the Cisco CUCM offline and the iD808 registered to a Cisco SRST if the iD808 is
resynchronised the Cisco configuration is lost and the iD808 cannot register to the SRST (F0021528)
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SIP Interface Versions in Version 1.400.5.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.5.0
- Client-side conference extended to support a 11-way conference
- Registration mechanism made multi-threaded
Defects Resolved in Version 1.400.5.0
- User Interface locked-up after making bulk changes to group permissions (F0021151)
- When enabling UI logs "All" the UI can lock up for several minutes (F0021324)
- Possible UI crash during a resync (F0021337)
- Sometimes when starting up with iE801s with the Ethernet ports enabled and Ethernet
cables connected the iD808 will not start up correctly and the DSPs get continuously
reset (F0021386)
Known Defects/Issues in Version 1.400.5.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
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SIP Interface Versions in Version 1.400.4.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.4.0
- Enhancements to improve memory useage during live updates and when executing system commands
Defects Resolved in Version 1.400.4.0
- The unit can run out of memory with a very large corporate directory. Now supports up to
60,000 sub-entries (F0020776)
- With an outbound group call on the selected handset pressing the conference key puts the group
call on hold but the on-hold tone is sent out to the group call (on-hold tone should only be sent
for point-to-point calls) (F0021002)
- Changing the "Intercom Talk Latching" setting on the iD808 UI does not send the new setting to iCMS (F0021020)
- When making an intercom group call from a speaker channel (that is seizing the appearance by
pressing the speaker channel key) the group call announce tone is not transmitted at the start
of the call (F0021038)
- When making an intercom group call from a speaker channel that has latching disabled (that is
seizing the appearance by pressing the speaker channel key) the speaker channel is left latched
on at the start of the call (F0021070)
- If the DHCP response is received close to the DHCP timeout period it is possible for the iD808
to report the DHCP status as "Bad" even though the DHCP response has been received (F0021072)
- When receiving a locked mixing group call, the audio may not be received (F0021103)
- Misleading wording on an informational popup seen during a failure of a call transfer attempt (F0021132)
- When the iD808 logs are generated errors are produced in the messages log file when attempting
to save the values of Array type CM variables such as the MAC address (F0021140)
- When sending CDR events the iD808 IP address and MAC address are obtained from opening network
sockets which is inefficient (F0021141)
- When making an intercom call from the handset with the handset muted the microphone is not muted
when the call is answered (F0021162)
- When using a profile with a large number of appearances an error can be produced in the
/var/log/messages file when starting up stating that the appearance index used is out of range (F0021163)
- When a mixing group call is on the handsfree interface and the handset free microphone is set
to gooseneck (or gooseneck exclusive) when the * key is pressed to talk back on the group the
gooseneck microphone indicator is not illuminated (F0021164)
- Lockup when receiving simultaneous point-to-point intercom and group calls (F0021170)
- On-hold calls displayed on dynamic keys may move from one dynamic key to another when answering
a call on a call apperance (F0021187)
- A call may fails when redialling a SIP URL from the redial menu (F0021190)
- When seizing an appearance on an Avaya Aura 6.2 PBX, if the number is not dialled within about
60 seconds the SIP session will fail which will cause the outbound call attempt to fail and
in addition there will be no indication of the line seize being timed out by the PBX (F0021191)
- iD808 stops showing "LOGGING ACTIVE" after re-powering it but iManager still shows it as
the diagnostics are still active (F0021206)
- IP addresses in the status screens are all listed as full 15 characters e.g. 010.001.032.020
which can make it difficult to read. The NTP server address in the Network Status screen is not
listed in this format (F0021208)
- An iD808 may crash while running an autoscript making intercom calls (F0021241)
- When making a call, two instances of the call are added to the call register (F0021244)
- MTFIF crashes when the originator of a call is being transferred by another iD808 when both are
using Cisco SRST (F0021260)
- Calls fails after making a large number of failed to connect intercom calls (F0021303)
- When a page is made read-only the menu item to delete paginating keys is disabled when viewing
the page. When going to a page that is not read-only then the menu item is now enabled.
This is wrong as paginating keys are fixed keys and should be independent of the page a user is viewing (F0021312)
- If an intercom line on a speaker channel is seized and the speaker page is changed the
speaker ID is not correctly updated causing unpredictable behaviour and the flashing handset
LED will not be cleared (F0021315)
Known Defects/Issues in Version 1.400.4.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
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SIP Interface Versions in Version 1.400.3.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.3.0
- No features or enhancements added to this release
Defects Resolved in Version 1.400.3.0
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- When a device has an intercom appearance and no telephony appearances the SIP Server Status
screen reports "No lines to register" (F0020917)
- The wording of section 4 of the readme file for installing a complete filesystem could be
improved (F0020939)
- Early media is not used for inter-site intercom calls (F0020952)
- When there is an intercom appearance on a fixed key and there is no Intercom Dial Number
programmed in iManager the appearance key is not displayed and the iCMS status icon
is yellow (F0020954)
- When there is a speed dial of type intercom or group and there is no Intercom Dial Number
programmed in iManager the speed dial gets converted to use the default appearance during
initialisation (F0020955)
- When a telephony appearance key is inactive and showing a red icon because it is trying to connect
to a backup server, a live update of the key incorrectly clears the inactive state
returning the icon to grey (F0020960)
- When an Avaya telephony appearance is not registered and the appearance key is moved,
an iCMS error "No valid master appearance found" may be incorrectly logged (F0020961)
- When the iD808 is trying to connect to the backup server for an Avaya PBX and the last
Avaya call appearance is removed from the keys on the device the "Connecting to backup
server" message remains on the device even though the device is no longer trying to
register to the Avaya PBX (F0020962)
- Incorrect CDR events are sent when moving group calls between the intercom handsfree
interface and the handset (F0020963)
- There is one way audio for P2P Intercom calls originated from the SB534 (F0020964)
Known Defects/Issues in Version 1.400.3.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The unit can run out of memory with a very large corporate directory (F0020776)
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SIP Interface Versions in Version 1.400.2.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.2.0
- Group talk member muting
- CDR connection status reported in the network status screen
Defects Resolved in Version 1.400.2.0
- When trying to make a Cisco call private when there is no conferencing resource
assigned within CUCM causes the call to fail with no user warning (F0016090)
- Screen not drawn correctly after a power up of the turret (F0018597)
- Changing a corporate directory non-default sub-entry that is used on a speed dial does not update
the key with the changes (F0020601)
- The voice recording tone on the iTurret keeps on beeping if you have it enabled and you dial an
invalid number (F0020637)
- When attempting to transfer an Asterisk call if the user aborts the transfer, the far end ID is
removed from the appearance key (F0020747)
- iCMS reports an error when a speaker channel is moved with key finder (F0020748)
- With Cisco and one privacy device and a private call on hold, a second call can also be made private (F0020749)
- A locally generated group call announce tone is louder than an inband played announce tone (F0020750)
- When a telephony call is ringing on a speaker channel an assign-assign cancels the ringing completely (F0020752)
- During a Cisco adhoc conference with Cisco not the default appearance, when adding in additional
participants the conference icon disappears (F0020753)
- When the limit of a Cisco adhoc conference is reached and the user attempts to add another Cisco
user the new call is dropped with a status code 0x26 which is not handled (F0020754)
- When there is an ARD call in a conference, and the conference call is in conference hold and the
ARD call is cleared from the far end, the status keys are not updated (F0020755)
- A live update of an intercom dial number where none were previously configured does not
update the appearance keys (F0020756)
- With a hoot on a speaker channel with four other active talkers and tap-latching enabled, after
attempting to talk when one of the active talker drops out the first speaker press does not activate
the talk (F0020757)
- With a hoot on a speaker channel and speaker source set to handset 1, when attempting to talk with
the talker limit reached transmission does not automatically start when an active talker drops out (F0020758)
- When receiving & from iCMS it may not be converted to a & (F0020759)
- A live update to remove a handset from the recording sources does not stop the recording warning
tone being played (F0020760)
- With speaker source set to handset 1 and the two appearance keys attached to speaker channels
and an active intercom call with talk latched on, when the i key is pressed the user is warned
that the handset is busy but then the talk on the existing intercom is unlatched (F0020761)
- When group talk (non-latching) is active and an incoming telephony call is answered on the handset,
the LEDs of other partner speakers in the group talk do not flash (F0020762)
- When an incoming telephony call which is part of group talk on a speaker channel is answered on
the selected handset with the speaker source is set to selected handset, pressing the group talk
key only changes the selected handset and does not action the group talk (F0020763)
- When a device is logged out it may still display the "Connecting to backup server" message (F0020764)
- With an Avaya call appearance on a speaker channel and the appearance in the busy elsewhere
state and shown on a dynamic key, barging in from the dynamic key does not clear the call off
the dynamic key (F0020765)
- With an Avaya bridge call appearance on a float key, the dynamic key showing the appearance
as busy elsewhere does not show the / number (F0020766)
- With a ringing intercom appearance on a float key, pressing the appearance key shows the handsfree
busy warning message and does not answer the call on the handset (F0020767)
- From iManager, when moving an intercom appearance that is also on a speaker channel from an unavailble
fixed key (shown on a float key) to a paginating key some errors are generated, the iCMS icon goes
yellow and the appearance is not added and deleted from the speaker channel (synchronising fixes the
problem) (F0020768)
- When two devices dial the same group call at the same time one user makes the call successfully but
the other user gets locked in a listen only state and some errors are produced. If the /2 appearance
is on a speaker channel then the user cannot assign the call to the handset and get the "Action
not possible" error message (F0020769)
- When an intercom appearance is on a speaker channel and the i key is pressed to seize a line,
dialling an invalid intercom number from the directories does not show the call in the failed
state. It changes from seized/connecting to idle (F0020770)
- Making a group call on a speaker channel with latching disabled leaves the speaker channel
latched when the call is connected (F0020771)
- Receiving an intercom P2P call to a speaker channel that is part of an active group talk with
a member of the group on the handset incorrectly activates the microphone for the connected
P2P call (F0020772)
- Receiving a mixing intercom group call to a speaker channel that is part of an active group does
not activate the microphone when the group call becomes connected or update the LED correctly to
show it is talking (F0020773)
- Various CDR issues with intercom calls (F0020775)
- When a function key is pressed and the key display changed to the other state, the transition to the
other state can be messy (F0020777)
- With intercom privacy enabled and the intercom appearance on a speaker channel and the selected
handset idle with handsfree selected and inbound intercom call partially disables the handsfree
mode of the handset (F0020778)
- With intercom privacy enabled and an active point-to-point intercom call, another inbound
point-to-point call results in a 180 ringing being generated before the call is rejected (F0020779)
- When an inbound group call is answered on a speaker channel and moved to a handset and then to
handsfree the appearance key displays a blank group call ID (F0020780)
- With an inbound mixing group that has been put on hold and then taken off hold, if the user tries
to talk the talk is not successful (F0020781)
- When using G722 coding, sometimes noise is heard at the far end as handsfree is turned on and
off (F0020782)
- With an intercom apperance on a speaker channel and dialing a logged out user the call fails
and leaves the speaker channel playing the ringback tone (F0020806)
- When receiving an ARD call and selecting 'Mute Alerts Now' , the mute alerts state does not clear
if the call is cancelled at the far end of the call (it does clear if cancelled locally) (F0020841)
- If there is an intercom group call on a speaker channel with paging enabled and the speaker
page is changed to one where the intercom appearance is not linked to a speaker channel
then the group call is cleared and the next highest priority group call is received (F0020891)
- If the iD808 is synchronised with a group call active the call is not shown to the user when
they log in (F0020906)
- If two users dial the same group number at the same time one user will join the group call
as a listener but the splash screen will be hidden (F0020908)
- With an intercom point to point call on the handsfree and a ringing group call. Pressing the
appearance or dynamic key answers the group call on the handset but when the point to point
call is cleared the group call is moved to the splash screen and is still shown on the
handset (F0020909)
- The nav key auto repeat timer is not cancelled when the informational popup is displayed which
can cause the menu to get stuck paging without any keys being pressed (F0020910)
- Intercom group call can still be received even when the SIP server status icon is flashing red.
If the iCS MCC server is connected the SIP server status icon should be yellow and it should
only be red if all lines have failed to register and the iCMS MCC server is not
connected (F0020911)
Known Defects/Issues in Version 1.400.2.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- The unit can run out of memory with a very large corporate directory (F0020776)
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SIP Interface Versions in Version 1.400.1.0
- Avaya Interface Version 1.40
- Cisco Interface Version 1.40
New Features/Enhancements Added in Version 1.400.1.0
- Intercom support via an iCS server
- Intercom point-to-point calls
- Intercom standard group calls
- Intercom mixing group calls
- Intercom locked group calls
- Intercom group call priority
- Intercom appearance keys (one or two)
- Intercom privacy
- Intercom handsfree operation
- Intercom splash screen
- Intercom speaker channel and handset operation
- Intercom group call directory
- Intercom announcement volume configuration
- Intercom call register logging configuration
- Function keys (DND, intercom privacy & recording warning tone)
- Individual speed dial icons for different directory entry types
- Immediate transfer option supported for speed dial keys
- Handsfree microphone configuration (internal, gooseneck & gooseneck exclusive)
- Corporate directory enhanced to support intercom entries and to increase capacity to 15000 (60000 sub-entries)
- Personal directory enhanced to support intercom and group call entries
- Independent dial plans for multple PBXs
- Record option supported for hoot channels
- Avaya redundancy supported for up to three registrar addresses
- Voice recording redundancy supported for up to two iCDS servers and two voice recorders
- Enhanced acoustics
- Configurable F2 key for iE801 modules to support either "Program" or "Speaker Page"
- Configurable speaker page name
- Speaker page icon on screen B of the iD808
- Shortcut menu via the * key may be disabled to facilitate initiating outbound dialling with the * and # keys
- Prompt screen shown for assign operations
- Line matching for SIP calls enhanced to support E.164 numbering
- "Call info" moved to "Device Info" menu
- "Speaker Settings" moved to "Speaker Actions" menu
- Logged in user status added to "i cms Status" screen
- Diagnostics and logging may be controlled from iCMS
- Device resync menu option
- UI logging enhanced to support sub-system filtering
- SSH CM command enhanced to support networktrace, sendlogsall, deletescript, deleteallscripts & logging
- Diagnostic autoanswer enhanced to allow auto clear to be enabled or disabled
Defects Resolved in Version 1.400.1.0
- Adjusting the Echo Canceller Tail Length parameter (under DebugDsp) causes the DSP to crash (F0016523)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- Sendlogs does not include information on the backup servers for a Cisco PBX (F0017347)
- There are some memory leaks when processing a live update of the dialplan xml file, the speakers
xml files and the keys xml files (F0017374)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
- CDR events may not contain the correct data as a result of live updates (F0019193)
- Some live updates to network settings should but do not report that a restart/reboot is required for
these changes to take effect (F0019519)
- The iD808 may lock up with "Activating a config change please wait" after changing DHCP to static
from iCMS (F0019584)
- When an appearance is visible and starts to ring, after a few seconds their dynamic key may rings even
though it is set for "Hidden Calls Only" (F0019678)
- Very occassionally a cached key is not drawn correctly immediately after initialisation (F0019679)
- The timezone settings are incorrect for Russia (F0019701)
- A conference call can be partially cleared off a speaker channel when one leg of the call clears
from the far end (F0019709)
- Minor errors to the user settings and call settings help text (F0019757)
- When barging in to a call with an Avaya PBX the to and the from tags in the Replaces header are
reversed (F0020075)
- DTMF telephone events from an iD808 do not work on calls blast dialled from a VCM (F0020170)
- When iD808 has all the 600 lines active and receives a new SIP INVITE the call overwrites the
600th call data structure (F0020247)
- Crash possible during a live update immediately followed by a resynchronise (F0020297)
- The UI can crash when clearing a call off the handset that is part of an active group (F0020554)
Known Defects/Issues in Version 1.400.1.0
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- Changing a corporate directory non-default sub-entry that is used on a speed dial does not update
the key with the changes (F0020601)
- The voice recording tone on the iTurret keeps on beeping if you have it enabled and you dial an
invalid number (F0020637)
- When attempting to transfer an Asterisk call if the user aborts the transfer, the far end ID is
removed from the appearance key (F0020747)
- iCMS reports an error when a speaker channel is moved with key finder (F0020748)
- With Cisco and one privacy device and a private call on hold, a second call can also be made private (F0020749)
- A locally generated group call announce tone is louder than an inband played announce tone (F0020750)
- When a telephony call is ringing on a speaker channel an assign-assign cancels the ringing completely (F0020752)
- During a Cisco adhoc conference with Cisco not the default appearance, when adding in additional
participants the conference icon disappears (F0020753)
- When the limit of a Cisco adhoc conference is reached and the user attempts to add another Cisco
user the new call is dropped with a status code 0x26 which is not handled (F0020754)
- When there is an ARD call in a conference, and the conference call is in conference hold and the
ARD call is cleared from the far end, the status keys are not updated (F0020755)
- A live update of an intercom dial number where none were previously configured does not
update the appearance keys (F0020756)
- With a hoot on a speaker channel with four other active talkers and tap-latching enabled, after
attempting to talk when one of the active talker drops out the first speaker press does not activate
the talk (F0020757)
- With a hoot on a speaker channel and speaker source set to handset 1, when attempting to talk with
the talker limit reached transmission does not automatically start when an active talker drops out (F0020758)
- When receiving & from iCMS it may not be converted to a & (F0020759)
- A live update to remove a handset from the recording sources does not stop the recording warning
tone being played (F0020760)
- With speaker source set to handset 1 and the two appearance keys attached to speaker channels
and an active intercom call with talk latched on, when the i key is pressed the user is warned
that the handset is busy but then the talk on the existing intercom is unlatched (F0020761)
- When group talk (non-latching) is active and an incoming telephony call is answered on the handset,
the LEDs of other partner speakers in the group talk do not flash (F0020762)
- When an incoming telephony call which is part of group talk on a speaker channel is answered on
the selected handset with the speaker source is set to selected handset, pressing the group talk
key only changes the selected handset and does not action the group talk (F0020763)
- When a device is logged out it may still display the "Connecting to backup server" message (F0020764)
- With an Avaya call appearance on a speaker channel and the appearance in the busy elsewhere
state and shown on a dynamic key, barging in from the dynamic key does not clear the call off
the dynamic key (F0020765)
- With an Avaya bridge call appearance on a float key, the dynamic key showing the appearance
as busy elsewhere does not show the / number (F0020766)
- With a ringing intercom appearance on a float key, pressing the appearance key shows the handsfree
busy warning message and does not answer the call on the handset (F0020767)
- From iManager, when moving an intercom appearance that is also on a speaker channel from an unavailble
fixed key (shown on a float key) to a paginating key some errors are generated, the iCMS icon goes
yellow and the appearance is not added and deleted from the speaker channel (synchronising fixes the
problem) (F0020768)
- When two devices dial the same group call at the same time one user makes the call successfully but
the other user gets locked in a listen only state and some errors are produced. If the /2 appearance
is on a speaker channel then the user cannot assign the call to the handset and get the "Action
not possible" error message (F0020769)
- When an intercom appearance is on a speaker channel and the i key is pressed to seize a line,
dialling an invalid intercom number from the directories does not show the call in the failed
state. It changes from seized/connecting to idle (F0020770)
- Making a group call on a speaker channel with latching disabled leaves the speaker channel
latched when the call is connected (F0020771)
- Receiving an intercom P2P call to a speaker channel that is part of an active group talk with
a member of the group on the handset incorrectly activates the microphone for the connected
P2P call (F0020772)
- Receiving a mixing intercom group call to a speaker channel that is part of an active group does
not activate the microphone when the group call becomes connected or update the LED correctly to
show it is talking (F0020773)
- Various CDR issues with intercom calls (F0020775)
- The unit can run out of memory with a very large corporate directory (F0020776)
- When a function key is pressed and the key display changed to the other state, the transition to the
other state can be messy (F0020777)
- With intercom privacy enabled and the intercom appearance on a speaker channel and the selected
handset idle with handsfree selected and inbound intercom call partially disables the handsfree
mode of the handset (F0020778)
- With intercom privacy enabled and an active point-to-point intercom call, another inbound
point-to-point call results in a 180 ringing being generated before the call is rejected (F0020779)
- When an inbound group call is answered on a speaker channel and moved to a handset and then to
handsfree the appearance key displays a blank group call ID (F0020780)
- With an inbound mixing group that has been put on hold and then taken off hold, if the user tries
to talk the talk is not successful (F0020781)
- When using G722 coding, sometimes noise is heard at the far end as handsfree is turned on and off (F0020782)
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SIP Interface Versions in Version 1.310.7.0
- Avaya Interface Version 1.31
- Cisco Interface Version 1.31
New Features/Enhancements Added in Version 1.310.7.0
- No features or enhancements added to this release
Defects Resolved in Version 1.310.7.0
- On-hold calls displayed on dynamic keys may move from one dynamic key to another when answering
a call on a call apperance (F0021187)
- A call may fails when redialling a SIP URL from the redial menu (F0021190)
- When seizing an appearance on an Avaya Aura 6.2 PBX, if the number is not dialled within about
60 seconds the SIP session will fail which will cause the outbound call attempt to fail and in
addition there will be no indication of the line seize being timed out by the PBX (F0021191)
Known Defects/Issues in Version 1.310.7.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
- When an appearance is visible and starts to ring, after a few seconds their dynamic key may rings even
though it is set for "Hidden Calls Only" (F0019678)
- Very occassionally a cached key is not drawn correctly immediately after initialisation (F0019679)
- The timezone settings are incorrect for Russia (F0019701)
- A conference call can be partially cleared off a speaker channel when one leg of the call clears
from the far end (F0019709)
- When barging in to a call with an Avaya PBX the to and the from tags in the Replaces header are
reversed (F0020075)
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SIP Interface Versions in Version 1.310.6.0
- Avaya Interface Version 1.31
- Cisco Interface Version 1.31
New Features/Enhancements Added in Version 1.310.6.0
Defects Resolved in Version 1.310.6.0
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- The iD808 can crash when synchronising immediately after a live update (F0020297)
- The iD808 can crash when clearing a call off the handset that is part of an active group (F0020554)
Known Defects/Issues in Version 1.310.6.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
- When an appearance is visible and starts to ring, after a few seconds their dynamic key may rings even
though it is set for "Hidden Calls Only" (F0019678)
- Very occassionally a cached key is not drawn correctly immediately after initialisation (F0019679)
- The timezone settings are incorrect for Russia (F0019701)
- A conference call can be partially cleared off a speaker channel when one leg of the call clears
from the far end (F0019709)
- When barging in to a call with an Avaya PBX the to and the from tags in the Replaces header are
reversed (F0020075)
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SIP Interface Versions in Version 1.310.5.0
- Avaya Interface Version 1.31
- Cisco Interface Version 1.31
New Features/Enhancements Added in Version 1.310.5.0
- Shortcut menu via the * key disabled to facilitate initiating outbound dialling with the * and # keys
Defects Resolved in Version 1.310.5.0
- There is an approximate 15 minute delay before the CDR socket is dropped and reconnected in the
case where the socket is dropped silently from the far end (F0020121)
Known Defects/Issues in Version 1.310.5.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
- When an appearance is visible and starts to ring, after a few seconds their dynamic key may rings even
though it is set for "Hidden Calls Only" (F0019678)
- Very occassionally a cached key is not drawn correctly immediately after initialisation (F0019679)
- The timezone settings are incorrect for Russia (F0019701)
- A conference call can be partially cleared off a speaker channel when one leg of the call clears
from the far end (F0019709)
- When barging in to a call with an Avaya PBX the to and the from tags in the Replaces header are
reversed (F0020075)
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SIP Interface Versions in Version 1.310.4.0
- Avaya Interface Version 1.31
- Cisco Interface Version 1.31
New Features/Enhancements Added in Version 1.310.4.0
- No features or enhancements added to this release
Defects Resolved in Version 1.310.4.0
- The iD808 sends a # for a # key in the SIP message when dialling from the iD808 whereas RFC3986
specify that %23 should be sent in the SIP message (F0020109)
Known Defects/Issues in Version 1.310.4.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
- When an appearance is visible and starts to ring, after a few seconds their dynamic key may rings even
though it is set for "Hidden Calls Only" (F0019678)
- Very occassionally a cached key is not drawn correctly immediately after initialisation (F0019679)
- The timezone settings are incorrect for Russia (F0019701)
- A conference call can be partially cleared off a speaker channel when one leg of the call clears
from the far end (F0019709)
- When barging in to a call with an Avaya PBX the to and the from tags in the Replaces header are
reversed (F0020075)
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SIP Interface Versions in Version 1.310.3.0
- Avaya Interface Version 1.31
- Cisco Interface Version 1.31
New Features/Enhancements Added in Version 1.310.3.0
- Transmit gain offset configuration added
Defects Resolved in Version 1.310.3.0
- There can be one way voice on the next call after cancelling out of a transfer attempt
with an Alcatel PBX (F0019552)
- The caller ID is incorrect after transferring a call on an Alcatel PBX (F0019553)
- When using the iD808 with an Avaya PBX, the subscription can repeatively refresh at an interval of
approximately 5 seconds after the expiry of initial subscription (F0019607)
- With the recording warning tone enabled and then toggling between the handset and
handsfree the tone keeps on being generated each time handsfree is selected (F0019627)
- When a call is cleared with the recording warning tone enabled there is a short burst of
tone heard before the call is ended (F0019638)
Known Defects/Issues in Version 1.310.3.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
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SIP Interface Versions in Version 1.310.2.0
- Avaya Interface Version 1.31
- Cisco Interface Version 1.31
New Features/Enhancements Added in Version 1.310.2.0
- Read-only speaker channels enhanced to prohibit assigning and unassigning appearance keys
- Auto-answer configuration option added to speaker channels
- Voice recording warning tone support added
- Support added for new Alcatel PBX type
- Acoustic shock protection configuration added
- Inter-DSP communications enhanced to improve error resilience
Defects Resolved in Version 1.310.2.0
- Call Forwarding fails with an Alcatel PBX (F0018825)
- The insert option should be greyed out when there are no free keys to perform the
insert action (F0019348)
- In the "Edit alert profile" menu, changing the ringtone doesn't play the new tone (F0019431)
- When using the iD808 with an Avaya Aura 6.1 PBX, the subscription to the dialog package
may time out causing the iD808 to loose common lamping status for a short period of
time (F0019460)
Known Defects/Issues in Version 1.310.2.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
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Back to the Index
SIP Interface Versions in Version 1.310.1.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.310.1.0
- Insert option added to key finder
- Call Forwarding support may be disabled in iManager
- Alert profile volume range changed from 1-16 to 0-16
- Improvements to the Engineering Tools menu include support of Trace Route
Defects Resolved in Version 1.310.1.0
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When there is an active group call and speaker source is set to handset 1, handset 2 or selected
handset, if another MRD is added to a blank speaker during the call this speaker defaults to the
input of the gooseneck. This can be remedied by putting the speaker onto the spare handset, then
pressing clear, and then it will use the correct handset for input and join the group (F0018370)
- With speaker source set to selected handset and auto select idle handset enabled if a group talk
is actioned while a member of the group is already active on the selected handset the selected
handset is incorrectly changed to the idle handset (F0018372)
- The alert mode "play after x seconds" does not work when being called from a local Cisco shared
line (F0018381)
- With speaker source configured for one of the handset modes and the unit configured as
push-to-talk if a group call is made with the handset muted and immediately changed to
handsfree the handfree mic appears to be unmuted on the UI but is in reality is still muted (F0018414)
- When receiving two calls set to use different alert profiles with the first call received set with
a higher priority and to ring immediately and the second call set with a lower priority and to
ring after 5 seconds, if the first ringing call in cancelled within the first 5 seconds the
ring tone continues to play until the 5 seconds has expired and the other ring tone starts (F0018456)
- After a number of group calls and switching between handfree and handset, on occasions the audio
emits from the handsfree with the audio input from the handset, while the turret is set to
handset (F0018548)
- Incoming call from Alcatel PBX using source port other than 5060 are responding back to SIP ringing
on port 65535 (F0018585)
- If a live update removes the Cisco appearance being used for the bulk registration the registration
will fail until the unit is resynchronised (F0018623)
- The TTL value for telephony RTP streams cannot be configured to a value less than 120 (F0018668)
- When using handset 2 for a group talk the handset key displays one of the channel names instead
of 'Group Talk' (F0018672)
- The new scrolling mode introduced for the directory lists has not been replicated for other
lists such as the call register list (F0018680)
- Live updates of voice services do not update the key so label changes are not seen (F0018793)
- Configuring call forwarding on an iD808 when interworking with a Cisco PBX does not configure the
Cisco PBX with the call forwarding setting (F0018844)
- When processing a Live Update the iD808 may noticeably 'slow down' (F0018865)
- Live updates may fail when there is a mismatch between the iD808 firmware and the iManager
software (F0018867)
- If keys are moved in iManager the iD808 may not accept the change or may lose the association with
the speaker channel. The device will need to be synchronised for the update to work (F0019000)
- Sometimes the "Speaker source" menu can be greyed out when no speaker channels are active
stopping the user from changing the setting (F0019096)
- After a DSP reset the master volume always defaults to maximum and not where the volume knob is
positioned. The volume can be corrected by adjusting the master volume knob (F0019185)
Known Defects/Issues in Version 1.310.1.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- When using SIP Subscription for Avaya Lines the P-Preferred-Identity has two sip headers (F0019171)
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SIP Interface Versions in Version 1.301.7.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.7.0
- Enhancement to the autodiscovery client and the discovery application to report the states of the
network, i cms and SIP server status icons
- Enhancement to the DSP diagnostics to report the status of the communication links between
the DSPs and PowerPC and to support the HALT command
Defects Resolved in Version 1.301.7.0
- Following a crash on the DSP the iD808 does not recover to a fully working state (F0016524)
Known Defects/Issues in Version 1.301.7.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- When there is an active group call and speaker source is set to handset 1, handset 2 or selected
handset, if another MRD is added to a blank speaker during the call this speaker defaults to the
input of the gooseneck. This can be remedied by putting the speaker onto the spare handset, then
pressing clear, and then it will use the correct handset for input and join the group (F0018370)
- With speaker source set to selected handset and auto select idle handset enabled if a group talk
is actioned while a member of the group is already active on the selected handset the selected
handset is incorrectly changed to the idle handset (F0018372)
- The alert mode "play after x seconds" does not work when being called from a local Cisco shared
line (F0018381)
- With speaker source configured for one of the handset modes and the unit configured as
push-to-talk if a group call is made with the handset muted and immediately changed to
handsfree the handfree mic appears to be unmuted on the UI but is in reality is still muted (F0018414)
- When receiving two calls set to use different alert profiles with the first call received set with
a higher priority and to ring immediately and the second call set with a lower priority and to
ring after 5 seconds, if the first ringing call in cancelled within the first 5 seconds the
ring tone continues to play until the 5 seconds has expired and the other ring tone starts (F0018456)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- After a number of group calls and switching between handfree and handset, on occasions the audio
emits from the handsfree with the audio input from the handset, while the turret is set to
handset (F0018548)
- Incoming call from Alcatel PBX using source port other than 5060 are responding back to SIP ringing
on port 65535 (F0018585)
- If a live update removes the Cisco appearance being used for the bulk registration the registration
will fail until the unit is resynchronised (F0018623)
- The TTL value for telephony RTP streams cannot be configured to a value less than 120 (F0018668)
- When using handset 2 for a group talk the handset key displays one of the channel names instead
of 'Group Talk' (F0018672)
- The new scrolling mode introduced for the directory lists has not been replicated for other
lists such as the call register list (F0018680)
- Live updates of voice services do not update the key so label changes are not seen (F0018793)
- Configuring call forwarding on an iD808 when interworking with a Cisco PBX does not configure the
Cisco PBX with the call forwarding setting (F0018844)
- When processing a Live Update the iD808 may noticeably 'slow down' (F0018865)
- Live updates may fail when there is a mismatch between the iD808 firmware and the iManager
software (F0018867)
- If keys are moved in iManager the iD808 may not accept the change or may lose the association with
the speaker channel. The device will need to be synchronised for the update to work (F0019000)
- Sometimes the "Speaker source" menu can be greyed out when no speaker channels are active
stopping the user from changing the setting (F0019096)
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SIP Interface Versions in Version 1.301.6.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.6.0
- No features or enhancements added to this release
Defects Resolved in Version 1.301.6.0
- When accessing an IVR system and pressing a DTMF digit that results in the IVR system redirecting
the media it is possible for the DTMF tone to remain constantly playing on the handset (F0018599)
- When selecting an idle ARD call on a speaker channel the user cannot hear the far end until the speaker
channel is unlatched (F0018999)
Known Defects/Issues in Version 1.301.6.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- When there is an active group call and speaker source is set to handset 1, handset 2 or selected
handset, if another MRD is added to a blank speaker during the call this speaker defaults to the
input of the gooseneck. This can be remedied by putting the speaker onto the spare handset, then
pressing clear, and then it will use the correct handset for input and join the group (F0018370)
- With speaker source set to selected handset and auto select idle handset enabled if a group talk
is actioned while a member of the group is already active on the selected handset the selected
handset is incorrectly changed to the idle handset (F0018372)
- The alert mode "play after x seconds" does not work when being called from a local Cisco shared
line (F0018381)
- With speaker source configured for one of the handset modes and the unit configured as
push-to-talk if a group call is made with the handset muted and immediately changed to
handsfree the handfree mic appears to be unmuted on the UI but is in reality is still muted (F0018414)
- When receiving two calls set to use different alert profiles with the first call received set with
a higher priority and to ring immediately and the second call set with a lower priority and to
ring after 5 seconds, if the first ringing call in cancelled within the first 5 seconds the
ring tone continues to play until the 5 seconds has expired and the other ring tone starts (F0018456)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- After a number of group calls and switching between handfree and handset, on occasions the audio
emits from the handsfree with the audio input from the handset, while the turret is set to
handset (F0018548)
- Incoming call from Alcatel PBX using source port other than 5060 are responding back to SIP ringing
on port 65535 (F0018585)
- If a live update removes the Cisco appearance being used for the bulk registration the registration
will fail until the unit is resynchronised (F0018623)
- The TTL value for telephony RTP streams cannot be configured to a value less than 120 (F0018668)
- When using handset 2 for a group talk the handset key displays one of the channel names instead
of 'Group Talk' (F0018672)
- The new scrolling mode introduced for the directory lists has not been replicated for other
lists such as the call register list (F0018680)
- Live updates of voice services do not update the key so label changes are not seen (F0018793)
- Configuring call forwarding on an iD808 when interworking with a Cisco PBX does not configure the
Cisco PBX with the call forwarding setting (F0018844)
- When processing a Live Update the iD808 may noticeably 'slow down' (F0018865)
- Live updates may fail when there is a mismatch between the iD808 firmware and the iManager
software (F0018867)
- If keys are moved in iManager the iD808 may not accept the change or may lose the association with
the speaker channel. The device will need to be synchronised for the update to work (F0019000)
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SIP Interface Versions in Version 1.301.5.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.5.0
- New configuration option added to ignore MRD ring signal when busy-elsewhere
Defects Resolved in Version 1.301.5.0
- The menu can be locked even when auto-hide is disabled (F0018512)
- When a group call of MRDs is made, with the speaker source set to handset 1, handset 2 or
selected handset, and signal is pressed only the last MRD selected in the group will ring,
not the rest (F0018514)
- An iTurret with iE801s may power up with the voice services being incorrectly allocated
to iE801s even though the iE801 Ethernet ports had been disabled (F0018519)
- Possible UI lockup after rapidly selecting group talk followed by clear (F0018520)
- Errors are generated in the error log when repeatedly toggling the handset mute switch
before and during a ringing incoming call, then answering the call on the dynamic key (F0018544)
- When the speaker source is set to selected handset, rapidly selecting and clearing groups
keys can result in audio links in the DSP getting mixed causing one way voice (F0018555)
- Possible one-way voice for an MRD call when using iE801 modules (F0018556)
Known Defects/Issues in Version 1.301.5.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- When there is an active group call and speaker source is set to handset 1, handset 2 or selected
handset, if another MRD is added to a blank speaker during the call this speaker defaults to the
input of the gooseneck. This can be remedied by putting the speaker onto the spare handset, then
pressing clear, and then it will use the correct handset for input and join the group (F0018370)
- With speaker source set to selected handset and auto select idle handset enabled if a group talk
is actioned while a member of the group is already active on the selected handset the selected
handset is incorrectly changed to the idle handset (F0018372)
- The alert mode "play after x seconds" does not work when being called from a local Cisco shared
line (F0018381)
- With speaker source configured for one of the handset modes and the unit configured as
push-to-talk if a group call is made with the handset muted and immediately changed to
handsfree the handfree mic appears to be unmuted on the UI but is in reality is still muted (F0018414)
- When receiving two calls set to use different alert profiles with the first call received set with
a higher priority and to ring immediately and the second call set with a lower priority and to
ring after 5 seconds, if the first ringing call in cancelled within the first 5 seconds the
ring tone continues to play until the 5 seconds has expired and the other ring tone starts (F0018456)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
- After a number of group calls and switching between handfree and handset, on occasions the audio
emits from the handsfree with the audio input from the handset, while the turret is set to
handset (F0018548)
- Incoming call from Alcatel PBX using source port other than 5060 are responding back to SIP ringing
on port 65535 (F0018585)
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SIP Interface Versions in Version 1.301.4.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.4.0
- Exit from menu by holding down the back key speeded up
- Fast exit from directory search supported
- Right justify text for right-handed keys supported
- Enhanced scrolling in the directories supported
- Auto-hide menu supported
- Auto-clear supported
- Paginating dynamic keys supported
- Busy-elsewhere/on-hold and busy-elsewhere-only on dynamic keys supported
- Shortcut to Speaker Page menu supported
- Alternative LED indicator scheme supported
- MAC addresses reported in Show Network screen
Defects Resolved in Version 1.301.4.0
- No defects resolved in this release
Known Defects/Issues in Version 1.301.4.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- When there is an active group call and speaker source is set to handset 1, handset 2 or selected
handset, if another MRD is added to a blank speaker during the call this speaker defaults to the
input of the gooseneck. This can be remedied by putting the speaker onto the spare handset, then
pressing clear, and then it will use the correct handset for input and join the group (F0018370)
- With speaker source set to selected handset and auto select idle handset enabled if a group talk
is actioned while a member of the group is already active on the selected handset the selected
handset is incorrectly changed to the idle handset (F0018372)
- The alert mode "play after x seconds" does not work when being called from a local Cisco shared
line (F0018381)
- With speaker source configured for one of the handset modes and the unit configured as
push-to-talk if a group call is made with the handset muted and immediately changed to
handsfree the handfree mic appears to be unmuted on the UI but is in reality is still muted (F0018414)
- When receiving two calls set to use different alert profiles with the first call received set with
a higher priority and to ring immediately and the second call set with a lower priority and to
ring after 5 seconds, if the first ringing call in cancelled within the first 5 seconds the
ring tone continues to play until the 5 seconds has expired and the other ring tone starts (F0018456)
- It is not possible to delete a paginating key that has a shortcut to menu on the key using key
finder (F0018470)
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SIP Interface Versions in Version 1.301.3.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.3.0
- No features or enhancements added to this release
Defects Resolved in Version 1.301.3.0
- Some dialled numbers on an Avaya PBX do not go to connected state and hence transmit audio
and DTMF tones do not work (F0018433)
Known Defects/Issues in Version 1.301.3.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
- When there is an active group call and speaker source is set to handset 1, handset 2 or selected
handset, if another MRD is added to a blank speaker during the call this speaker defaults to the
input of the gooseneck. This can be remedied by putting the speaker onto the spare handset, then
pressing clear, and then it will use the correct handset for input and join the group (F0018370)
- With speaker source set to selected handset and auto select idle handset enabled if a group talk
is actioned while a member of the group is already active on the selected handset the selected
handset is incorrectly changed to the idle handset (F0018372)
- The alert mode "play after x seconds" does not work when being called from a local Cisco shared
line (F0018381)
- With speaker source configured for one of the handset modes and the unit configured as
push-to-talk if a group call is made with the handset muted and immediately changed to
handsfree the handfree mic appears to be unmuted on the UI but is in reality is still muted (F0018414)
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SIP Interface Versions in Version 1.301.2.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.2.0
- Support added for a new iManager user configuration option to allow a group talk action to not
connect MRD channels that are in the busy elsewhere state
- Support added for Samsung K9F2G08U0C-SCB0 flash devices
Defects Resolved in Version 1.301.2.0
- The network status screen and associated icon can report a DSP reset after an upgrade when using iE801
modules (F0016364)
- When making an outgoing Cisco call to a number that is setup to forward calls to a PSTN number the
UI shows the call as being immediately connected and two superimposed ringback tones are heard.
When the call is answered at the far end a ringback tone is still played to the user until the call
ends (F0017929)
- An Avaya PW does not work correctly after a live update to the appearance. A resync resolves the issue (F0017993)
- There is a noticeable delay from pressing the group select button to actively transmitting
voice on all channels for a large SbRTP group (F0018096)
- The UI may crash if it receives a corrupt corporate directory where some entries include labels or
addresses with embedded line feed characters (F0018165)
- When a VPW is selected in the call failed state but with the appearance data indicating the line is
busy elsewhere the barge in attempt fails and the appearance data is lost and hence the user cannot
barge into the call (F0018183)
- Live update of the low bandwidth / standard SbRTP setting does not restart the SbRTP
link and hence a resync is required to restore operation (F0018202)
- There may be one-way voice on a SIP call when using iE801 modules (F0018217)
- The DebugDsp IGMP status command for the iE801 modules only works every other time (F0018225)
- When clearing down a barged in MRD line a burst of noise may be heard on the unit clearing
down (F0018229)
- When configured for speaker source handset 1 or 2 or selected handset and handset press to talk if
the user presses a MRD speaker channel and then rapidly the talk button on the handset the handset
is not unmuted although it looks like it is unmuted from the UI (F0018231)
- There may be no transmit audio if an MRD call is cleared and then rapidly connected again (F0018233)
- When using the CM sendlogs command the user has to wait for the lcc_status.txt to timeout (F0018237)
- It is possible to get a call register profile error after a repower (F0018253)
- When a user answers an incoming MRD call, all other users continue to see a ringing indicator for
several seconds (F0018265)
- When configured for speaker source selected handset it is not possible to select a group talk (F0018266)
- When configured for DHCP if the IP address changes there is no indication to the user that a problem
has occurred and the device does not auto announce to i cms so iManager may still be using
the old address (F0018281)
- The error message "ERROR:UI_kf_key_assigned:../src/UI_states.c: key finder class 4 not handled" can
be generated in the log file (F0018301)
- Some calls may not be connected correctly to the microphone or speaker as a result of incorrectly
selecting the wrong mixer (F0018343)
- When the latching mode is set for push-to-latch and the speaker source is set to handset 1 or handset 2,
active group calls cannot be cleared off the handset by pressing the handset clear key. They can only
be cleared by pressing the group talk key (F0018364)
- When the speaker source is set to selected handset, auto assign to speaker channel (by just selecting
the speaker channel to assign a call on the handset and not pressing assign followed by the speaker
channel) doesn't assign the call (F0018367)
Known Defects/Issues in Version 1.301.2.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- When upgrading iD808s with iE801 modules fitted and their Ethernet ports enabled the iE801s can
occasionally report the DHCP status as "Bad" (F0018332)
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SIP Interface Versions in Version 1.301.1.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.301.1.0
- No features or enhancements added to this release
Defects Resolved in Version 1.301.1.0
- MRDs with both global muting enabled and using low bandwidth SbRTP generate a click to other
local users when a user initiates a call (F0018201)
- The iE801 does not rejoin its IGMP groups after removing and reconnecting its Ethernet cable (F0018224)
- The MRD initial ring option sends back a ring signal to the calling end when answering a call (F0018226)
- UI crash when the user has accessed to a large number of telephony lines and the total number of
appearances for those lines exceeds 600 (F0018227)
- There is no media when connecting additional calls and it is not possible to barge into some SbRTP
calls when the total number of links including all VR streams, SbRTP channels and telephony calls
exceed 24 (F0018228)
- There can be audio distortion on the iE801 modules when the iD808 DSP is processing a large number
of channels e.g. 24 SbRTP channels (F0018230)
Known Defects/Issues in Version 1.301.1.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The network status screen and associated icon can report a DSP reset after an upgrade when using iE801
modules (F0016364)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- When making an outgoing Cisco call to a number that is setup to forward calls to a PSTN number the
UI shows the call as being immediately connected and two superimposed ringback tones are heard.
When the call is answered at the far end a ringback tone is still played to the user until the call
ends (F0017929)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- An Avaya PW does not work correctly after a live update to the appearance. A resync resolves the issue (F0017993)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
- iE801 voice recording streams may not work correctly immediately after an upgrade (F0018039)
- There is a noticeable delay from pressing the group select button to actively transmitting
voice on all channels for a large SbRTP group (F0018096)
- The UI may crash if it receives a corrupt corporate directory where some entries include labels or
addresses with embedded line feed characters (F0018165)
- Live update of the low bandwidth / standard SbRTP setting does not restart the SbRTP
link and hence a resync is required to restore operation (F0018202)
- There may be one-way voice on a SIP call when using iE801 modules (F0018217)
- The DebugDsp IGMP status command for the iE801 modules only works every other time (F0018225)
- When clearing down a barged in MRD line a burst of noise is heard on the unit clearing down. This
was observed when using low bandwidth 4ms SbRTP with global muting on. (F0018229)
- When configured for speaker source handset 1 or 2 or selected handset and handset press to talk if
the user presses a MRD speaker channel and then rapidly the talk button on the handset the handset
is not unmuted although it looks like it is unmuted from the UI (F0018231)
- There may be no transmit audio if an MRD call is cleared and then rapidly connected again (F0018233)
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SIP Interface Versions in Version 1.300.19.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.19.0
- Additional audio environment added named Trader 2 which is identical to Trader 1 but with
squelch disabled (CN2753)
- Enhancements to sendlogs to include the lcc_status.txt status file
Defects Resolved in Version 1.300.19.0
- Infrequent UI crash associated with displaying the animated ringing icons (F0017236)
- Occassionally an Avaya appearance can remain as busy-elsewhere on an Avaya Aura 6.0 PBX
even though the call is cleared down (F0017316)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
- If a Cisco appearance gets incorrectly stuck on busy elsewhere because of network errors a
failed attempt to barge in to the call does not clear the common lamping back to idle (F0017491)
- When logging out, unsubscription messages that require authentication are not handled
resulting in lines still being subscribed (F0017502)
- The iD808 can generate error messages in the engineering messages log file during startup
or during a resync (F0017515)
- The iD808 can fail to turn call forward on / off if the last appearance of the master appearance
is busy (F0017534)
- The iCMS status icon can be yellow with an error stating that a DTD did not match after a
live update when there is a mismatch between the i cms version and the iD808
firmware (F0017565)
- An iD808 voice recording stream can contain improper audio after a voice recording configuration
change (F0017574)
- Cisco AdHoc conference is not used when you conference in a call that was established prior to pressing
the conf key. This also is true when you try to conference an on hold call that was established prior
to pressing conf button (F0017583)
- In certain circumstances a Cisco AdHoc Conference can not be initialised. No warning message is
given from the UI (F0017639)
- When the 600th SbRTP appearance is added to a user profile or permissions to the 600th one, an
error is produced saying that the maximum user appearances have been used. This error should
only be generated when the 601th is attempted to be added (F0017689)
- When running two "Send Logs" from the menu soon after each other the DSP status will not contain
all of the information as the "~" character at the end of the previous log will be shown at the
start of the new log causing the new log to be ended too early (F0017690)
- If the PBX does not provide a remote identity it is not possible to barge in to a Cisco call (F0017705)
- It is not possible to configure more than one Avaya PW per PBX (F0017727)
- When configured with DHCP disabled the iE801 subnet mask and default gateway cannot be set via
the UI (F0017766)
- On occasions the second iE801 attached to a iD808 will boot up too slowly after a repower, and
as a result the DSP resources on this iE801 will not be used, therefore only allowing 400 of the
600 MRDs to be used. Another re-power solves this (F0017791)
- In certain circumstances if you have a iD808 with two iE801s connected, the iE801s may end up having
the same IP address as each other (F0017797)
- Cisco Adhoc conferencing fails if the PBX does not send a 183 Session Progress message (F0017800)
- G.729 or G.722 codecs may stop processing packets on all transmit or all receive streams (F0017804)
- When creating a Cisco AdHoc conference, if the first call (the call put on hold) is created on an iE801
then when the conference is created the user cannot talk or receive from the conference (F0017835)
- The UI may lockup when changing the SbRTP VAD setting with a live update with a large number of
SbRTP channels (600) on the device (F0017837)
- When changing the SbRTP VAD setting with a live update and channels which cannot be assigned DSP
resources disappear from the device until the next resync (F0017839)
- If a turret places a Cisco AdHoc conference on Hold, another turret is unable to pick that conference
up from hold. A SIP 403 forbidden is received (F0017851)
- When using iE801s and calls are shared between the iD808 and iE801s, if a conference is created and
a far end user clears one of the calls any remaining calls might lose the microphone and speaker
connection (F0017854)
- When using two iE801s and the calls are spread across the iD808 and both iE801s in a conference all
of the calls are not linked correctly resulting in some conference members not receiving all the
audio from all the other conference members (F0017855)
- Occasionally an MRD can remain constantly ringing after the ring signal has finished (F0017868)
- It is possible for the user to select the wrong appearance when pressing a nav key immediately followed
by a paginating appearance key (F0017897)
- When dialling a busy number on Cisco without voice mail set up and the call fails the warning
"Action not possible" is shown (F0017898)
- iE801 Ethernet interfaces may not work if an iD808 is upgraded to v1.30 from v1.1x or earlier (F0018019)
Known Defects/Issues in Version 1.300.19.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When performing a live update of a large number of SbRTP channels on a device fitted with iE801
modules then the SbRTP channels allocated to the iE801 may not work correctly until a resync
or repower (F0017779)
- For voice recording streams, RTCP packets are still sent when Recording Enabled is unticked
if the RTCP Enabled box is ticked (F0017988)
- An Avaya PW does not work correctly after a live update to the appearance. A resync resolves the issue (F0017993)
- i cms requires only the SIP registrar field to be completed when creating a new PBX.
However the iD808 will not attempt to register when only the registrar field is populated (F0018005)
- When barging in to a Cisco call with no Caller ID and then afterwards dropping out of the
call, the call remains showing the conference label even though it is now back to a
point-to-point call (F0018006)
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SIP Interface Versions in Version 1.300.18.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.18.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.18.0
- Occasionally attempts to conference calls using Cisco Ad-Hoc conferencing fail (F0017704)
- UI crash when attempting to make a Cisco call private when the call has no remote caller ID (F0017725)
- Infrequent UI crash when seating a user (F0017758)
- With no iE801 modules and more than 200 SbRTP channels configured on the device all channels appear to
be active but only 200 channels will be active (F0017780)
- When using static IP addresses if an iE801 is configured with an IP address of 0.0.0.0 no error is
reported (F0017783)
Known Defects/Issues in Version 1.300.18.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- Infrequent UI crash associated with displaying the animated ringing icons (F0017236)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- Occassionally an Avaya appearance can remain as busy-elsewhere on an Avaya Aura 6.0 PBX
even though call is cleared down (F0017316)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
- If a Cisco appearance gets incorrectly stuck on busy elsewhere because of network errors a
failed attempt to barge in to the call does not clear the common lamping back to idle (F0017491)
- When logging out, unsubscription messages that require authentication are not handled
resulting in lines still being subscribed (F0017502)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- The iD808 can generate error messages in the engineering messages log file during startup
or during a resync (F0017515)
- The iD808 can fail to turn call forward on / off if the last appearance of the master appearance
is busy (F0017534)
- An iD808 voice recording stream can contain improper audio after a voice recording configuration
change (F0017574)
- Cisco AdHoc conference is not used when you conference in a call that was established prior to pressing
the conf key. This also is true when you try to conference an on hold call that was established prior
to pressing conf button (F0017583)
- In certain circumstances a Cisco AdHoc Conference can not be initialised. No warning message is
given from the UI (F0017639)
- Cisco AdHoc Conferencing is not supported across Master and Expansion units as defined within
Cisco Call Manager (F0017640)
- When the 600th SbRTP appearance is added to a user profile or permissions to the 600th one, an
error is produced saying that the maximum user appearances have been used. This error should
only be generated when the 601th is attempted to be added (F0017689)
- When running two "Send Logs" from the menu soon after each other the DSP status will not contain
all of the information as the "~" character at the end of the previous log will be shown at the
start of the new log causing the new log to be ended too early (F0017690)
- When the iE801 network ports are disabled. The dsp_status file in the "Send Logs" does not get
completed and gets stuck. The DSP continues to function and can send and receive messages (F0017691)
- If the PBX does not provide a remote identity it is not possible to barge in to a Cisco call (F0017705)
- Incorrect Caller ID displayed when transferring a Cisco call with no Caller ID (F0017723)
- Small memory leak when dialling out a telephony call (F0017765)
- Outbound dialling UI slows down after running a specific client-side conference autoscript for a
considerable time (F0017778)
- When configuring a large number of SbRTP channels on a device fitted with iE801 modules then the
SbRTP channels allocated to the iE801 may not work correctly until a resync or repower (F0017779)
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SIP Interface Versions in Version 1.300.17.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.17.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.17.0
- With Cisco ad-hoc conferencing and "Advanced Ad Hoc Conference Enabled" set to false in the
Cisco CM if a participant of an ad-hoc conference who is not the originator tries to add to
the conference the request get rejected but when the user returns to the conference the call
state icon displays the conference icon instead of the call established icon (F0017409)
- With conferencing calls, if one call is set up on the iD808 and the other is set up on an iE801
the handsfree may not work correctly as the change is being sent to the wrong DSP resulting in
handsfree not working or the handset also being connected (F0017525)
- When using Cisco, sometimes the label of a busy appearance can show the remote address (or
directory match) and not the "Conference" label when other users barge in or the appearance
is put on hold and taken off hold (F0017560)
- When starting up, voice recorder streams from any iE801 modules may not work correctly.
Synchronising or modifying the configuration corrects the problem. (F0017562)
Known Defects/Issues in Version 1.300.17.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- Infrequent UI crash associated with displaying the animated ringing icons (F0017236)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- Occassionally an Avaya appearance can remain as busy-elsewhere on an Avaya Aura 6.0 PBX
even though call is cleared down (F0017316)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
- If a Cisco appearance gets incorrectly stuck on busy elsewhere because of network errors a
failed attempt to barge in to the call does not clear the common lamping back to idle (F0017491)
- When logging out, unsubscription messages that require authentication are not handled
resulting in lines still being subscribed (F0017502)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- The iD808 can generate error messages in the engineering messages log file during startup
or during a resync (F0017515)
- The iD808 can fail to turn call forward on / off if the last appearance of the master appearance
is busy (F0017534)
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SIP Interface Versions in Version 1.300.16.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.16.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.16.0
- iD808 and iE801 volume controls have a large initial jump which means that low volumes
cannot be achieved when the master volume control is set to maximum (F0017510)
- Cisco unattended transfers leaves a ringback tone on the handset after the transfer is
completed (F0017511)
- The iD808 can fail to turn call forward on / off on an Avaya Aura 6.0 PBX (partial fix for F0017534)
Known Defects/Issues in Version 1.300.16.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- Infrequent UI crash associated with displaying the animated ringing icons (F0017236)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- Occassionally an Avaya appearance can remain as busy-elsewhere on an Avaya Aura 6.0 PBX
even though call is cleared down (F0017316)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- With Cisco ad-hoc conferencing and "Advanced Ad Hoc Conference Enabled" set to false in the
Cisco CM if a participant of an ad-hoc conference who is not the originator tries to add to
the conference the request get rejected but when the user returns to the conference the call
state icon displays the conference icon instead of the call established icon (F0017409)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
- If a Cisco appearance gets incorrectly stuck on busy elsewhere because of network errors a
failed attempt to barge in to the call does not clear the common lamping back to idle (F0017491)
- When logging out, unsubscription messages that require authentication are not handled
resulting in lines still being subscribed (F0017502)
- The iE801 Ethernet ports can remain disabled when disabling and enabling the Ethernet
ports on the module (F0017514)
- The iD808 can generate error messages in the engineering messages log file during startup
or during a resync (F0017515)
- With conferencing calls, if one call is set up on the iD808 and the other is set up on an iE801
the handsfree may not work correctly as the change is being sent to the wrong DSP resulting in
handsfree not working or the handset also being connected (F0017525)
- The iD808 can fail to turn call forward on / off if the last appearance of the master appearance
is busy (F0017534)
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SIP Interface Versions in Version 1.300.15.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.15.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.15.0
- Possible UI crash when hosting a conference if it involves a call of any type that resides on
an iE801 speaker channel (F0017462)
Known Defects/Issues in Version 1.300.15.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- With Cisco ad-hoc conferencing and "Advanced Ad Hoc Conference Enabled" set to false in the
Cisco CM if a participant of an ad-hoc conference who is not the originator tries to add to
the conference the request get rejected but when the user returns to the conference the call
state icon displays the conference icon instead of the call established icon (F0017409)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
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SIP Interface Versions in Version 1.300.14.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.14.0
- Configuration option for the MRD ring signal to be automatically sent to the line when
initially connecting to the call on the handset from an idle state (CN2745)
Defects Resolved in Version 1.300.14.0
- No defects resolved in this release
Known Defects/Issues in Version 1.300.14.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- With Cisco ad-hoc conferencing and "Advanced Ad Hoc Conference Enabled" set to false in the
Cisco CM if a participant of an ad-hoc conference who is not the originator tries to add to
the conference the request get rejected but when the user returns to the conference the call
state icon displays the conference icon instead of the call established icon (F0017409)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
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SIP Interface Versions in Version 1.300.13.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.13.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.13.0
- The iE801#2 right speaker does not change volume when the master volume is adjusted (F0017420)
- The iD808 can generate a constant engaged tone on the handset or speaker when dialling a busy
extension via a MGCP gateway to another Cisco Call Manager. The tone can only be cleared by
seizing a new line and if on handsfree, pressing handsfree (F0017432)
Known Defects/Issues in Version 1.300.13.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
- All ring tones seem to change tone when the volume changes from 11 to 12 (F0017338)
- The "Orphaned" label is not displayed on handset 2 when a call on handset 2 is initially orphaned (F0017390)
- With Cisco ad-hoc conferencing and "Advanced Ad Hoc Conference Enabled" set to false in the
Cisco CM if a participant of an ad-hoc conference who is not the originator tries to add to
the conference the request get rejected but when the user returns to the conference the call
state icon displays the conference icon instead of the call established icon (F0017409)
- When privacy is enabled during a G729 call, or a G729 call is barged into on Avaya 6, a
loud burst of static-like noise is emitted from the turret (F0017424)
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SIP Interface Versions in Version 1.300.12.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.12.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.12.0
- DTMF tone can be left playing on the handset after the call is cleared down (F0017359)
- UI crash after making a change to a PBX setting following on from deleting an entire PBX (F0017360)
- DTMF tones and telephone events don't work for SIP calls established on an iE801 module.
This only applies to iE801s with network cables connected (F0017364)
- When using Cisco call manager 8, incoming calls may always use the /1 number as the PKID is
not correctly checked (F0017383)
- Possible MTFIF crash caused by overwriting allocated memory when sending a refer message (F0017384)
Known Defects/Issues in Version 1.300.12.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
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SIP Interface Versions in Version 1.300.11.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.11.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.11.0
- Attended transfer on an Avaya Aura 6.0 can fail if the user transferring the call talks for
at least a minute on the second leg of the transfer before completing the transfer (F0017349)
Known Defects/Issues in Version 1.300.11.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
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SIP Interface Versions in Version 1.300.10.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.10.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.10.0
- Cisco ad-hoc conferencing does not work when no privacy appearances are defined (F0017277)
- On Cisco when joining a participant to an ad-hoc conference using the redial menu the conference
is created on another call appearance (F0017278)
- When the initiator creates a 3-way Cisco ad-hoc conference, if one of the remote devices leaves,
the initiator still shows the call as a conference and not a point-to-point call (F0017280)
Known Defects/Issues in Version 1.300.10.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
- The i cms status icon stays yellow after adding appearance keys and setting the default
appearance for the first time (F0017275)
- The user cannot make any SIP call after removing all the call appearances and then adding
back a call appearance without doing a resync (F0017279)
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SIP Interface Versions in Version 1.300.9.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.9.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.9.0
- When an ARD call is made or answered and is immediately cleared then sometimes the ring back tone
continues to sound on the speaker despite having been cleared completely off the module (F0016805)
- UI crash when changing the "adhoc conferencing" in i manager (F0017243)
- DTMF tones are not transmitted for calls that are set up on an iE801 (F0017255)
Known Defects/Issues in Version 1.300.9.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
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SIP Interface Versions in Version 1.300.8.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.8.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.8.0
- If a call on a speaker channel is dipped or muted and put on hold at the far end then the
speaker channel will not become un-dipped and remain at the dipped level (F0016319)
- An iD808 does not send an announce to the i cms server when the 'Use DNS to locate
iCMS Server' setting is changed (F0017194)
- A live update of a PBX entry can cause registration issues (F0017231)
- DTMF digits are not recognised with Aura Media Messaging, when the trunk type is set
to SIP (F0017234)
Known Defects/Issues in Version 1.300.8.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
- The handsfree volume is very loud and distorted when an ARD is ringing on the handset when
the volume set to maximum (F0017218)
- When using Avaya Aura Messaging, fast double presses of the same DTMF digit are not
recognised. Only the first key press is recognised (F0017235)
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SIP Interface Versions in Version 1.300.7.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.7.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.7.0
- The Cisco escape code when a conference is not possible is not handled (F0017187)
Known Defects/Issues in Version 1.300.7.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
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SIP Interface Versions in Version 1.300.6.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.6.0
- Cisco ad-hoc conferencing (CN2730)
- "mem" key event added to the autoscript function to generate a UI memory report log message
Defects Resolved in Version 1.300.6.0
- A live update of a corporate or personal directory entry does not update any speed dials using
the entry (F0016937)
- There is a memory leak when the device makes multiple calls which can eventually crash the
device (F0016957)
- Changing the directories via i manager with a live update can crash the iD808 (F0017026)
- The iD808 can incorrectly report a Cisco registration error if the device starts up without access
to the PBX and then gets access to the PBX (F0017037)
- Deleting call appearances via i manager with a live update can crash the iD808 (F0017044 & F0017168)
- There is a 500 internal server error generated by an Avaya PBX when cancelling call forwarding on
Avaya call appearances (F0017059)
- Cisco appearances can stay in the BUSY state when trying to access voicemail without the voicemail
address being configured in i manager (F0017070)
- Speed dials can be cached incorrectly when added via i manager when the directory entry
is also added (F0017121)
- Minor issue where the date format in the Device Info screen could be ambiguous (F0017180)
Known Defects/Issues in Version 1.300.6.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- The iD808 can generate error messages in the engineering messages log file if a telephony
call gets cleared down by both the local and remote end at approximately the same time (F0017105)
- The iD808 can crash when cancelling out of the second leg of a transfer when the PBX is
configured incorrectly (F0017173)
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SIP Interface Versions in Version 1.300.5.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.5.0
- No features or enhancements added to this release
Defects Resolved in Version 1.300.5.0
- Occassionally hoot or MRD speaker channels do not work after a resync and show a red icon when
selected (F0016348)
- Using an Avaya PBX with the registrar address configured as an IP address and the SIP domain
as a domain string, the second leg of a transfer SIP INVITE is sent to the wrong IP address. (F0016463)
- Hoots on speaker channels do not connect (F0016519)
- When a call has privacy enabled and then put on hold the call still shows the padlock, however on
Cisco PBX once a call is on hold the call is not private thus allowing other to barge into that
call, hence the padlock should be grey and not gold (F0016729)
- The side tone does not work when answering a VPW (F0016752)
- Problem where the call appearances end up showing as "busy elsewhere" even though they are not (F0016862)
- Cannot always transfer a call when using a Cisco PBX (F0016914)
- When making a speed dialled call that contains a pause character if the user switches handsfree mode
it is possible for a DTMF tone to remain playing continuously on the handset earpiece or the
handsfree speaker (F0016915)
- Disabling the iE801 network ports from start up does not work correctly and if a network cable
is connected then the iE801 will request a DHCP IP address if enabled. The IP address is also
reported to i cms. If no network cable is connected the the default 192.168.1.253/4 is
reported to i cms. If the network ports are disabled then a blank IP address should be
reported to i cms (F0016938)
- A device can locked up with the "Activating a configuration change and restarting unit please wait..."
screen after making changes to the VLAN settings from i cms, repower to clear (F0016963)
- The CDR conference on hold event has the should record flag as 1 but it should be 0 (F0016968)
Known Defects/Issues in Version 1.300.5.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
- A live update of a corporate or personal directory entry does not update any speed dials using the
entry. Synchronising fixes the issue (F0016937)
- There is a memory leak when the device makes multiple calls (F0016957)
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SIP Interface Versions in Version 1.300.4.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.4.0
- Cisco meet-me conferencing (CN2715)
- "Unassign Speaker" menu item replaced by "Speaker Actions" menu (CN2715)
- "Program Group" menu item moved to "Speaker Actions" menu (CN2715)
- "Speaker Page" menu item moved to "Speaker Actions" menu (CN2715)
- iE801 speaker keys "Ring", "Program" & "Mute" renamed to "F1", "F2" & "F3" (CN2715)
- iE801 "Function Key 1" configuration added to iE801 settings menu (CN2715)
- "Ring" softkey renamed as "Signal" soft key (CN2715)
- Auto assign line to speaker channel added (CN2715)
- Speaker Page label added to iE801 header bar (CN2715)
- Speaker Page icon enhanced (CN2715)
- Speaker channel icons enhanced (CN2715)
- Speaker channel LED indicators enhanced (CN2715)
- Handset muted icons enhanced (CN2715)
- Improved audio settings for the iE801
Defects Resolved in Version 1.300.4.0
- When changing the iE801 Ethernet port settings the DSP can crash (F0016507)
- When an active call is both on a speaker channel and the selected handset and the user presses
assign followed by the speaker channel, the call is correctly moved to the speaker channel but
pressing any dial pad keys does not enter outbound dialling mode (F0016607)
- Script files for autoscript cannot be copied to the device as SBEngineer (F0016619)
- SIP registration failure caused by a SIP stack DNS issue where the domain is not
updated when changed in resolv.conf causing DNS queries to fail (F0016622)
- A SIP call can failed to be added to a dynamic key when put on hold by another user (F0016626)
- When interworking with an Avaya Aura 6.0 PBX the first voice mail message received by the
iD808 illuminates the message waiting indicator but the second voice mail message received
turns the message waiting indicator off again (F0016633)
- The outbound dialling prefix feature does not work if the number contains white space (F0016641)
- Auto hold does not work if the user has auto select idle handset enabled but only one handset
on the device (F0016647)
- SbRTP calls do not work with 1ms or 2ms packets with the low bandwidth setting (F0016648)
- With VAD enabled a clicking noise can be heard for SbRTP calls (F0016649)
- During an authenticated call if a SIP message from an iD808 is challenged by a PBX the iD808
repeats the SIP message with a second NONCE value added to the message instead of replacing
the existing NONCE message (F0016653)
- If the second leg of a call becomes a conference then the transfer cannot be completed (F0016676)
- The iD808 can crash if a large amount of speaker updates are performed (F0016685)
- With Cisco CUCM 8.0 the iD808 remains in the connected state when dialling a busy PSTN
number (F0016687)
- The second leg of a transfer does not go to the connected state if a SIP 180 ringing message
is not received before the SIP 200 OK and hence the transfer cannot be completed (F0016694)
- The side tone does not work when answering a VPW (F0016752)
Known Defects/Issues in Version 1.300.4.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- When a call has privacy enabled and then put on hold the call still shows the padlock, however on
Cisco PBX once a call is on hold the call is not private thus allowing other to barge into that
call (F0016729)
- Occassionally hoot or MRD speaker channels do not work after a resync and show a red icon when
selected (F0016348)
- Using an Avaya PBX with the registrar address configured as an IP address and the SIP domain
as a domain string, the second leg of a transfer SIP INVITE is sent to the wrong IP address. (F0016463)
- The SNMP packet count reporting does not include packets that have been filtered out by the
Blackfin DSP (F0016772)
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SIP Interface Versions in Version 1.300.3.0
- Avaya Interface Version 1.30
- Cisco Interface Version 1.30
New Features/Enhancements Added in Version 1.300.3.0
- The "Show Versions" menu now reports the overall "Firmware Version" and the Avaya and Cisco
Interface versions (CN2715)
- Support added to signal ARDs which are on-hold allowing other devices to monitor this
and display the ARD channel as on-hold-elsewhere and if not private add the call to the
dynamic keys (CN2715)
- Added the ability to transmit DTMF tones on SIP calls which are in a conference hosted by the
device. This feature can be enabled or disabled via i cms (default is disabled) (CN2715)
- Support added to configure pauses in speed dials to allow them to contain a dial string
followed by a list of DTMF digits to be played when the call connects. Pauses can be entered as
‘p’ or ‘P’. The first ‘p’ splits the dial string from the DTMF tones and additional ‘p’(s)
add a 1 second delay to the tones. The character ‘P’ has been added to the list of symbols
under the * key when in numeric entry mode. This type of dialling is supported in directory
dial, speed dial, VPWs, call register/redial and voice mail server dial (CN2715)
Defects Resolved in Version 1.300.3.0
- VPWs should not be converted to speed dials in the event of a profile update error. This can cause a
crash. (F0016403)
- The DSP command IGMP can take up to 12 seconds to print out all of the information which means some of
it is lost when called from the DALL command (F0016431)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
- Adding new appearances via a live update does not work correctly (F0016485, F0016486 & F0016515)
- An outbound call attempt for an Avaya ARD configured for Hotkey dialling out of E1/T1
interface fails shortly after the far end starts to ring (F0016504)
- The "should_record" flag in the CDR event CDR_CALL_HANGUP is set to "1" instead of "0" (F0016506)
- The gooseneck microphone gain is set 12dBs too high compared to version 1.200.13.0 with the result that
the echo canceller performance is poor (F0016525)
- When the device has been running for a long time, error messages can be seen in the log files as a message ID
counter overflows (F0016538)
- Cisco registration sometimes fails to register with the server when using a name string as the registrar address.
A resync does not fix the issue but restarting the device does (F0016541)
- The DSCP value is not set correctly in RTP/SbRTP packets when set to a non-zero value (F0016561)
- Common lamping SIP NOTIFY messages from a Cisco PBX are rejected if the "To" header contains an extension
number instead of a PKID number even when the PKID number exists in the Request URI header (F0016572)
- Using Cisco CUCM 8.0 an incorrectly dialled number should result in an audible error message being played.
The iD808 does not play this message unless the handsfree button is pressed (F0016579)
- The UI crashed when permissions to a Cisco appearance were removed in iManager while the device was logged
in and then the device was synchronised soon afterwards (F0016580)
Known Defects/Issues in Version 1.300.3.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- Occassionally hoot or MRD speaker channels do not work after a resync and show a red icon when selected (F0016348)
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New Features/Enhancements Added in Version 1.300.2.0
- CDR changes to match "Speakerbus 3rd party Interface Specification Issue 13"
Defects Resolved in Version 1.300.2.0
- SbRTP links can start up with the receive disabled causing one way voice until the device is
synchronised (F0016136)
- In the CDR output the record flag is a 1 when it should be a 0 for the CALL_INCOMING_AUDIO_STOP with a
barged in to an ARD on a speaker (F0016164)
- When processing a large live update from i cms the UI can become unresponsive (F0016301)
- When assigning an orphaned call to the handset the style of the speaker channel is copied to the handset (F0016303)
- The sound levels on the iE801 units connected to the iD808 are excessively loud (F0016316)
- The audible ringing indication can be distorted on iD808 units with iE801s attached (F0016318)
- Occasionally a live update is lost (F0016330,F0016366,F0016379,F0016403)
- Out-of-sync configuration error when editing a speaker channel in i cms to an appearance already on
another speaker channel caused by the iD808 incorrectly sending an update to i cms (F0016344)
- When changing the colour of a telephony appearance the style is only updated for active calls so idle
SIP appearances are not updated and also appearances on fixed keys are not checked if they are on a
speaker channel (F0016347)
- Enabling the iE801 network ports doesn't restart the DHCP timer so DHCP is never successful (F0016354)
- Live update of keys that should be on float keys does not work (F0016355)
- Sometimes the iD808 can start up with a yellow network icon reporting that the device needs to be
repowered to activate a configuration change. In this state the i cms and SIP icons are red (F0016356)
- A reset occurred after sending log files (F0016379)
- When changing the microphone connection to calls (e.g. by moving from a speaker channel to a handset)
the microphone link to other calls can be broken (F0016390)
- The UI can crash or the updates not correctly processed when doing a live update of the voice recorder,
pbx or dial plan xml files (F0016396)
Known Defects/Issues in Version 1.300.2.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- Occassionally hoot or MRD speaker channels do not work after a resync and show a red icon when selected (F0016348)
- VPWs should not be converted to speed dials in the event of a profile update error. This can cause a crash (F0016403)
- The DSP command IGMP can take up to 12 seconds to print out all of the information which means some if
it is lost when called from the DALL command (F0016431)
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New Features/Enhancements Added in Version 1.300.1.0
- Support for one or two iE801 expansion units
- Support for live updates from i cms
- Support for redundant servers when interoperating with a Cisco PBX
- Support for automated testing (auto-script, autoanswer and auto-record)
- Early media support added to SIP telephony calls
- Cisco registrar addresses extracted from Cisco config files to support load balancing
- SIP domain and SIP registrar addresses use separated
- When completing an add, delete, modify, move or view in key finder the user is returned to the
key finder selector instead of the main menu
- Assign-assign enhanced to allow hoots and MRDs to be easily unassigned from speaker channels
- ‘Codecs in use’ and ‘Packet size’ information added to the Call Info page
Defects Resolved in Version 1.300.1.0
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- When barging in to a Cisco adhoc conference call the call register shows a bxxxxxxxx label and any attempt
to dial from the call register fails (F0016106)
- There can be a brief audio burst heard on a iD808 speaker when another local user starts talking on a SbRTP
channel with global muting enabled (F0016181)
- The SNMP does not report any "interface" stats for the MIB-II support (F0016246)
- Deleted entries from the personal directory are only reported to i cms if they were used for speed dials (F0016273)
- The iD808 can lose communications with i cms when network settings are being changed (F0016296)
Known Defects/Issues in Version 1.300.1.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- When processing a large live update from i cms the UI can become unresponsive (F0016301)
- When assigning an orphaned call to the handset the style of the speaker channel is copied to the handset (F0016303)
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New Features/Enhancements Added in Version 1.210.2.0
- No features or enhancements added to this release
Defects Resolved in Version 1.210.2.0
- The iD808 cannot support 14 simultaneous G729/G722 calls. For example, when more than 6 G729 calls
are connected the DSP runs out of processing time and resets (F0016522)
- The DSCP value is not set correctly in RTP/SbRTP packets when set to a non-zero value (F0016561)
Known Defects/Issues in Version 1.210.2.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- SbRTP links can start up with the receive disabled causing one way voice until the device is
synchronised (F0016136)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
- An outbound call attempt for an Avaya ARD configured for Hotkey dialling out of E1/T1
interface fails shortly after the far end starts to ring (F0016504)
- Common lamping SIP NOTIFY messages from a Cisco PBX are rejected if the "To" header contains an extension
number instead of a PKID number even when the PKID number exists in the Request URI header (F0016572)
- Using Cisco CUCM 8.0 an incorrectly dialled number should result in an audible error message being played.
The iD808 does not play this message unless the handsfree button is pressed (F0016579)
- Auto hold does not work if the user has auto select idle handset but only one handset on the device (F0016647)
- SbRTP calls do not work with 1ms or 2ms packets with the low bandwidth setting (F0016648)
- With VAD enabled a clicking noise can be heard for SbRTP calls (F0016649)
- A transfer cannot be completed if the second leg becomes a conference by another user barging in (F0016676)
- With Cisco CUCM 8.0 the iD808 remains in the connected state when dialling a busy PSTN number (F0016687)
- The second leg of a transfer does not go to the connected state if a SIP 180 ringing message is not received
before the SIP 200 OK and hence the transfer cannot be completed (F0016694)
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New Features/Enhancements Added in Version 1.210.1.0
- Added the ability to transmit DTMF tones on SIP calls which are in a conference hosted by the
device
- Support added to configure pauses in speed dials to allow them to contain a dial string
followed by a list of DTMF digits to be played when the call connects. Pauses can be entered as
‘p’ or ‘P’. The first ‘p’ splits the dial string from the DTMF tones and additional ‘p’(s)
add a 1 second delay to the tones. The character ‘P’ has been added to the list of symbols
under the * key when in numeric entry mode. This type of dialling is supported in directory
dial, speed dial, VPWs and call register/redial
Defects Resolved in Version 1.210.1.0
- No defects resolved in this release
Known Defects/Issues in Version 1.210.1.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- SbRTP links can start up with the receive disabled causing one way voice until the device is
synchronised (F0016136)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
- The iD808 cannot support 14 simultaneous G729/G722 calls. For example, when more than 6 G729 calls
are connected the DSP runs out of processing time and resets (F0016522)
- The DSCP value is not set correctly in RTP/SbRTP packets when set to a non-zero value (F0016561)
- Auto hold does not work if the user has auto select idle handset but only one handset on the device (F0016647)
- SbRTP calls do not work with 1ms or 2ms packets with the low bandwidth setting (F0016648)
- With VAD enabled a clicking noise can be heard for SbRTP calls (F0016649)
- A transfer cannot be completed if the second leg becomes a conference by another user barging in (F0016676)
- With Cisco CUCM 8.0 the iD808 remains in the connected state when dialling a busy PSTN number (F0016687)
- The second leg of a transfer does not go to the connected state if a SIP 180 ringing message is not received
before the SIP 200 OK and hence the transfer cannot be completed (F0016694)
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SIP Interface Versions in Version 1.200.17.0
- Avaya Interface Version 1.20
- Cisco Interface Version 1.21
New Features/Enhancements Added in Version 1.200.17.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.17.0
- SbRTP links on speaker channels can start up incorrectly causing them to not connect when selected.
This bug was introduced in version 1.200.16.0 (F0016519)
Known Defects/Issues in Version 1.200.17.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
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SIP Interface Versions in Version 1.200.16.0
- Avaya Interface Version 1.20
- Cisco Interface Version 1.21
New Features/Enhancements Added in Version 1.200.16.0
- Cisco meet-me conferencing (CN2715)
- DTMF tones may now be transmitted during a conference hosted locally by the iD808 (CN2715)
- Pauses may be included in the dial strings for directory entries, speed dials & VPWs (CN2715)
Defects Resolved in Version 1.200.16.0
- SbRTP links can start up with the receive disabled causing one way voice until the device is
synchronised (F0016136)
- Auto hold does not work if the user has auto select idle handset but only one handset on the device (F0016647)
- SbRTP calls do not work with 1ms or 2ms packets with the low bandwidth setting (F0016648)
- With VAD enabled a clicking noise can be heard for SbRTP calls (F0016649)
- A transfer cannot be completed if the second leg becomes a conference by another user barging in (F0016676)
- With Cisco CUCM 8.0 the iD808 remains in the connected state when dialling a busy PSTN number (F0016687)
- The second leg of a transfer does not go to the connected state if a SIP 180 ringing message is not received
before the SIP 200 OK and hence the transfer cannot be completed (F0016694)
Known Defects/Issues in Version 1.200.16.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
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SIP Interface Versions in Version 1.200.15.0
- Avaya Interface Version 1.20
- Cisco Interface Version 1.20
New Features/Enhancements Added in Version 1.200.15.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.15.0
- The iD808 cannot support 14 simultaneous G729/G722 calls. For example, when more than 6 G729 calls
are connected the DSP runs out of processing time and resets (F0016522)
- The DSCP value is not set correctly in RTP/SbRTP packets when set to a non-zero value (F0016561)
Known Defects/Issues in Version 1.200.15.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
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SIP Interface Versions in Version 1.200.14.0
- Avaya Interface Version 1.20
- Cisco Interface Version 1.20
New Features/Enhancements Added in Version 1.200.14.0
- The "Show Versions" menu now reports the overall "Firmware Version" and the Avaya and Cisco
Interface versions
Defects Resolved in Version 1.200.14.0
- An outbound call attempt for an Avaya ARD configured for Hotkey dialling out of E1/T1
interface fails shortly after the far end starts to ring (F0016504)
- Common lamping SIP NOTIFY messages from a Cisco PBX are rejected if the "To" header contains an extension
number instead of a PKID number even when the PKID number exists in the Request URI header (F0016572)
- Using Cisco CUCM 8.0 an incorrectly dialled number should result in an audible error message being played.
The iD808 does not play this message unless the handsfree button is pressed (F0016579)
Known Defects/Issues in Version 1.200.14.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- The iD808 cannot support 14 simultaneous G729/G722 calls. For example, when more than 6 G729 calls
are connected the DSP runs out of processing time and resets (F0016522)
- When testing with Avaya Aura 6.0, the MWI does not illuminate when a voice mail message is available (F0016464)
- The DSCP value is not set correctly in RTP/SbRTP packets when set to a non-zero value (F0016561)
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New Features/Enhancements Added in Version 1.200.13.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.13.0
- There is a delay in setting the prime user indicator in the SbRTP control header for ringdown calls which
can result in a delay in activating global muting for other users (F0016181)
Known Defects/Issues in Version 1.200.13.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- The iD808 cannot support 14 simultaneous G729/G722 calls. For example, when more than 6 G729 calls
are connected the DSP runs out of processing time and resets (F0016522)
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New Features/Enhancements Added in Version 1.200.12.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.12.0
- No CDR event is sent when barging into an ARD call on a speaker channel (F0016138)
- If a voice recording stream is changed in iManager with no audio devices configured
the change is not applied until the device is repowered (F0016139)
- When barging into a SIP call on a speaker channel with latching disabled the talk stop event
can be sent before the answered by event (F0016140)
Known Defects/Issues in Version 1.200.12.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
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New Features/Enhancements Added in Version 1.200.11.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.11.0
- When VLAN is enabled on the iD808 the iD808 does not respond to an Autodiscovery request message
and hence will not be listed in the Autodiscovery application (F0016021)
- With the Avaya call manager configured for G729B, audio stops after about three seconds when
calling an Avaya SIP phone. When the call is cleared the DSP resets (F0016060)
- Users can barge into private lines on Cisco. This bug was introduced in version 1.200.10.0 (F0016061)
- When calling a busy Cisco extension that forwards to VoiceMail, the redial menu contains the
voicemail phone number instead of the dialled party number (F0016065)
Known Defects/Issues in Version 1.200.11.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
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New Features/Enhancements Added in Version 1.200.10.0
- Assign-assign enhanced to allow it to be used to unassign an idle appearance from a speaker channel (CD2701)
- Screen saver time out changed from 2 hours to 4 hours (CN2701)
Defects Resolved in Version 1.200.10.0
- Synchronising delay when changing between static and DHCP addressing from i cms (F0015981)
- With DND activated on an iD808, an incoming call on a Cisco call appearance is not shown (F0015988)
- When using the Free key wizard some entries cannot be selected (F0015991)
- iD808 goes into the connected state when dialling a busy Cisco extension. It should indicate that the
extension is busy (F0016003)
- There is a discrepancy between the CDR event specification and what the iD808 does for the record flag
within the on-hold message (F0016024)
- Internal log error messages generated when forwarding a Cisco call (F0016038)
Known Defects/Issues in Version 1.200.10.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance (F0015795)
- When using Cisco the caller ID does not get updated until the call is answered on a unattended
transfer (F0016020)
- When VLAN is enabled on the iD808 the iD808 does not respond to an Autodiscovery request message
and hence will not be listed in the Autodiscovery application (F0016021)
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New Features/Enhancements Added in Version 1.200.9.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.9.0
- Various minor CDR bugs (F0015879,F0015880,F0015884,F0015885)
- If a users barges into a telephony call, speaks and then leaves the call the VAD activity indicator is left
active and will only be updated if another user speaks and then goes silent (F0015890)
- The iD808 does not report an error if it is incorrectly configured with appearances from two different
Cisco PBXs. This is an invalid configuration. (F0015917)
- UI crash when editing a key during a resync (F0015925)
- When trying to barge into a private call, the handset finger & appearance finger do not display the
"red X" for very long and also the two full beeps on the handset are not played to indicate the call
has failed (F0015928)
- Attempting to call a Cisco "Caller ID Withheld" entry from the call register busy's out the appearance
for approximately 5 minutes (F0015929)
- No error is reported if there is a mismatch in configuration between Cisco Call Manager and iManager (F0015933)
- A key can be deleted from the iD808 and iManager when a change is saved away on the device at the
same time that the device is resynced from iManager (F0015949)
- When the handset volume softkey has been pressed and the handset volume bar is displayed DTMF tones
do not work. When the volume bar is cleared the DTMF tones work correctly again (F0015950)
- With auto handset mute enabled, calls on the handset get muted if SIP calls are answered on speaker
channels (F0015958)
Known Defects/Issues in Version 1.200.9.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance. (F0015795)
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New Features/Enhancements Added in Version 1.200.8.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.8.0
- Using Cisco, an iD808 can fail to cancel privacy leaving a grey padlock on the handset (F0015789)
- Sometimes a blank label is displayed in the call register (F0015790)
- If one user makes a Cisco call private at the same time as another user barges in to the call it
is possible to end up with a private conference call. In this situation it should be possible to
cancel the privacy but the user is blocked with the message that privacy is not allowed in a
conference. (F0015800)
- The expiry time for a Cisco barge-in attempt is too long (F0015849)
- If a Cisco privacy attempt fails because the barge-in attempt times out at the PBX, the next
privacy attempt immediately fails with Cisco reporting a 500 internal server error (F0015850)
- If two different users try to simultaneously seize the same Cisco line both line seizes can be
successful resulting in two separate calls existing against the appearance key (F0015865)
- The Speaker Source option is not selectable after the handset busy message is shown when attempting
to talk on a speaker channel with the speaker source set to handset 1 (F0015867)
- The iD808 crashes after barging in to a Cisco call that is on a speaker channel using the speaker
key while the speaker source is set to handset 1 (F0015868)
Known Defects/Issues in Version 1.200.8.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- With a Cisco appearance set as the default appearance and with Avaya appearances configured on the
device, when call forward is turned on the iD808 also sends the FNU's to Avaya to enable call
forwarding on the Avaya PBX. The call forwarding should only be enabled for the default appearance. (F0015795)
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New Features/Enhancements Added in Version 1.200.7.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.7.0
- When deleting an appearance that is attached to an on-hold call that is being displayed on a dynamic
key the on-hold call is not cleared off the dynamic key (F0015756)
- The iD808 can crash while deleting a group talk key (F0015757)
- When VLAN is enabled and a network trace is active any packets that have a frame size of 1518
(fragmented packets) are truncated to 1514 bytes and therefore will not be fully decoded by
Wireshark (F0015772)
- In the SIP Server Status screen reporting of Cisco registration is confusing to the user because it
is not clear that it is using bulk registration (F0015773)
- Cisco lines do not show busy-elsewhere when a shared line is seized at another unit (only shows
busy-elsewhere after the number has been dialled) (F0015774)
- The MWI indicator when used with Cisco PBX, will illuminate for a voice mail left on a bridged
appearance. However it will not illuminate if the call appearance is not on the "01" device for
the Cisco PBX (F0015778)
Known Defects/Issues in Version 1.200.7.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
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New Features/Enhancements Added in Version 1.200.6.0
- Serial interface and SSH passwords changed
Defects Resolved in Version 1.200.6.0
- iD808 sends "500 Internal Server error" when receiving Notify message with Message Waiting status for
the Cisco privacy devices (F0015657)
- Dynamic keys displaying ringing Cisco calls may not update correctly when the call ends (F0015661)
- A call can disappear off the handset when a user tries to make an Avaya call private if the call is
local to the PBX and the call is already private at the remote end. This problem only occurs when the
internal index of the appearance key is zero (F0015691)
- After answering a Cisco call, with privacy enabled for the iD808, the call appearance can show busy
elsewhere instead of established and putting the call on hold can result in both a on-hold-here and
on-hold-elsewhere call for the same appearance being displayed on two dynamic keys (F0015695)
- Speaker Source menu can be greyed out even when there are no speaker channels active (F0015703)
- "Handset busy" is displayed instead of "Action not possible" when pressing hold for a handset when
a speaker mic is active on the handset (F0015707)
- When changing the DHCP setting the device UI freezes and requires a repower before it can be logged
in (F0015716)
Known Defects/Issues in Version 1.200.6.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
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New Features/Enhancements Added in Version 1.200.5.0
- No features or enhancements added to this release
Defects Resolved in Version 1.200.5.0
- When making a Cisco unattended transfer the appearance of the first leg of the call remains busy elsewhere
or on-hold elsewhere unit the second leg is answered (F0015586)
- When a far end user drops out of a conference the appearance key does not update from conference to the
last user extension left in and active (F0015596)
- Entering a page title that only contains whitespace can cause an exception in i cms. When editing
a page title or alert name a check is made to ensure an empty string cannot be saved but this should be
changed to ensure titles containing only whitespace are also rejected (F0015616)
- Sometimes when Cisco calls are answered the call is added to the missed call log and not the received call
log causing the missed call icon to be displayed (F0015642)
- A continually failing registration can result in many of the registration attempts having the incorrect
details filled in e.g. Cisco extensions for non-Cisco registrations (F0015643)
- If the Audio Device setting is changed in the User Preferences menu and saved and the menu is exited
quickly (by repeatedly pressing the back key) then the change is lost (F0015648)
- Fragmented packets received on ports used for Cisco privacy notifications do not have the port number
changed by the DSP and so are rejected by the PowerPc when received (F0015649)
- Output audio settings cannot be adjusted on the DSP. All settings are fixed at their minimum values and
cannot be changed (F0015651)
Known Defects/Issues in Version 1.200.5.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
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New Features/Enhancements Added in Version 1.200.4.0
- Maximum number of SbRTP calls increased from 60 to 200 (CN2682)
- Low bandwidth SbRTP mode introduced. Configurable via i cms (CN2682)
- Updated status bar icons and red unavailable key icons
- CM variables added to monitor last read time of various message queues for
diagnostics purposes
- Send logs enhanced to include status of the message queues
- Send logs enhanced to include status of the NTP client
- Speed improvements for the upgrade process
- "Help Topics" changed to "Device Info" in the main menu
Defects Resolved in Version 1.200.4.0
- Screens can get into a strange state after cancelling a call at the same time as the screen saver is about
to be activated (F0013975)
- Ringback tone issue during an attended transfer of a call using the Cisco PBX (F0015228)
- On Cisco lines with user A connected to user B and user B performing an unattended
transfer to user C, user A should hear a ringback tone but doesn't (F0015230)
- An iD808 can erroneously show a yellow Cog and report the error "Failed to retrieve #03 config
file from Cisco TFTP server" (F0015233)
- An iD808 can erroneously report "Failed to retrieve config files from primary Cisco TFTP server"
after all Cisco lines have been deleted from the device (F0015235)
- Incorrect status reporting for the registration of Cisco lines (F0015237)
- When an Avaya call is put on hold and the device is synchronised the appearance shows
'seized elsewhere' and the call cannot be cleared (F0015239)
- When accessing the DebugDsp interface via the RS232 serial interface if the DALL command
is entered and then the return is pressed several times to queue up several DALL commands
the DspProc app can lock up when the message queue to DebugDsp becomes full and then the
iD808 needs to be repowered to operate correctly (F0015242)
- The gooseneck level indicator is not updated when the gooseneck microphone active indicator is
turned on. It is only updated when a change is made (F0015243)
- On Cisco lines with user A connected to user B and user B performing an unattended transfer to
user C, user A hear a local ringback tone in connected state but when you move it to speaker
channel via assign and to speaker channel the ringback tone is not heard (F0015250)
- If a fifth person barges into an ARD using an iD808 voice service appearance the call goes
to the handset and just rings until the fourth person drops out, then is added to the
call (F0015254)
- When an iD808 tries to barge into an MRD call on a handset when the talker limit is reached,
the user should not be able to talk on the call, which they cannot but they should be able to
hear the conference but they cannot (F0015256)
- Losing the network connection while in an Avaya call makes the call appearance in use
unable to make new calls (F0015257)
- The iD808 devices can suffer a UI lockup and fail to synchronise and reboot successfully
when placed within a subnet that has high SbRTP traffic (F0015266)
- Unattended transfer failure when performing the transfer quickly (F0015275)
- Updating of keys when in key finder is very slow when multiple MRD common lamping changes occur at
the same time (F0015277)
- Memory leak if the UI to MTFIF registration message is full and a registration message is
discarded (F0015278)
- UI can lock up when speaker paging or rapid speaker channel activity (F0015294)
- When an invalid number is dialled using the Cisco PBX the outgoing call attempt fails as
expected but the appearance key is incorrectly left in the busy elsewhere state (F0015302)
- When editing a speaker channel with speaker paging disabled the automatic level reduction
setting is always greyed out no matter which priority mode is selected however it can still
be selected when priority is not 1 (F0015303)
- When an iD808 is sending or receiving a lot of MRD traffic, making and receiving telephony calls
on the iD808 is very slow on the user interface (F0015309)
- When rapidly pressing a group talk key for a group of eight MRD channels it is possible for
the group talk to be on but none of the channels to be talking (F0015318)
- When rapidly pressing a group talk key for a group of eight MRD channels it is possible to show
the speaker active with microphone muted icon (F0015319)
- If a device is synchronised repeatedly the directories fail to load showing errors
"Phonebook not initialised"/"directory pre-load failed" and "Already preloaded"/"directory
pre-load failed" (F0015322)
- Very occassionally during an upgrade a key can be cached with a blank key image instead of its
real image. This has been observed for keys at the bottom of screen A (F0015323)
- After a period of several hours in a call audio can be lost in one direction and after a
further delay audio can be lost in the other direction (F0015329)
- When there is an & symbol in a page title it is not displayed correctly (F0015336)
- The MRD connected icon is left in the incorrect state when two MRDs are in a conference
both on speaker channels and both a member of an active speaker group. The problem
occurred when the conference is put on hold while the group talk is active. One
conference member shows held and the other shows connecting (F0015343)
- When changing from a speaker page with an active group talk containing two MRD channels
with speaker paging enabled and in the same conference, to a speaker page where those
two speaker channels are idle and then back to the original speaker page, only one of
the speaker channels are shown with the correct connected icon (F0015344)
- Call-forwarding-on-busy fails for Cisco and Asterisk calls if the first busy appearance
has an outgoing call (F0015350)
- An invalid appearance that is also on a speaker channel shows a red background for the audio
device icon but a grey background instead of red for the call state icon (F0015352)
- When an invalid appearance that is also on a speaker channel is deleted the speaker
channel gets cleared but still shows the red unavailable icons (F0015355)
- When the user has a dynamic key on the bottom key on screen A and there is a Cisco call
on hold on that key eventually it get updated as a busy elsewhere key even though the
call is still on-hold. The call may still be taken off-hold by selecting the key (F0015359)
- The microphone indicator is left in the active state when an ARD call that is active on
a speaker channel is assigned to a handset by pressing the appearance key while the menu
is active (F0015360)
- The Speaker Activity Indication Timeout cannot be set to its maximum value of 10000 ms
through i manager. Only values up to 5000 are accepted (F0015361)
- Minor errors with the online help text. In Key definition help "dynamic speaker channels"
should be "speaker channels". In Safe mode help "Show i cms" should use an italicised
i (F0015362)
- A connecting ARD with the talker limit reached and latching enabled cannot be unlatched (F0015363)
- Missing CDR event when answering an ARD call on a dynamic key with the menu active (F0015364)
- Missing CDR event (incoming audio start) when starting to talk on an ARD call
currently receiving audio with global muting enabled (F0015365)
- With two handsets and only one Avaya hidden appearance and with the selected handset in
the requesting privacy state and idle, selecting an Avaya appearance can cancel the
privacy request with a privacy warning message even though it is possible to make a
private call (F0015366)
- Mute Alerts Now can be deactivated by pressing clear on the other handset (F0015367)
- If a connecting ARD call is assigned to an idle speaker channel the ringback tone is
played out on the line. The ringback tone is not played if the ARD appearance was
already on the speaker channel (F0015368)
- When assigning an MRD or Hoot to an idle speaker channel that is part of an active
group it does not stay in the connected state but drops down to idle (F0015369)
- "Line to handset" functionality does not work for an MRD call currently in an active
group with one member already on a handset (F0015370)
- Moving calls around various speaker channels can cause i cms a problem as the same
speaker channel can appear twice in the speakers partial update. This can cause the
update to be lost (F0015371)
- CDR event "Incoming Audio Stop" for an idle or busy elsewhere call even though there
was no start event or any audio heard (F0015372)
- When editing a line with the appearance on a speaker channel the speaker channel is
cleared when the change is saved (F0015373)
- The U-boot "htest 30" command reports the wrong key name for switches SW23, SW24, SW25,
SW44, SW45 & SW46 (F0015376)
- Group Talk keys are not cached (F0015377)
- Intermittent UI crash when changing the device IP address via i cms (F0015383)
- An unattended transfer sometimes fails as a result of sending the SIP BYE too early (F0015390)
- With 200 ARDs and a single SIP line the SIP status icon is red after a synchronise
and only goes green after the next re-registration attempt (F0015391)
- An unattended transfer sometimes fails as a result of the SIP stack responding slowly (F0015400)
- If you press Conference by "Accident" pressing the 'cancel' key returns you to the point
to point call, however the Appearance and Handset finger are indicating 'Conference' therefore
you are unable to transfer the call (F0015442)
- When barging into a point-to-point call using Avaya there is a delay (of up to two minutes)
before the caller ID changes to CONFERENCE 2 (F0015449)
- When re-synchronising after adding or removing the number of telephone extensions that a
user has permission to use it is possible that the iD808 will attempt to register or
subscribe to additional appearances that it should not be registering or subscribing to (F0015470)
- When transferring a call to a number which isn't seated you get a error message "transfer
incomplete original call disconnected" which closes the call at one end but not the other
and leaves the call on hold (F0015471)
- DTMF tones are not cleared when moving a call from handset to handsfree or vice-versa (F0015473)
- ARD microphone stays active even when the speaker channel key is released causing the
microphone to be active when the far end answers but the microphone indicator does not show
the microphone as active (F0015474)
- The overall state is not correctly set when clearing the second leg of a transfer (F0015475)
- When the second leg of a transfer fails and the first and second legs are cleared the
warning box is always shown and not cleared (F0015476)
- The side tone is active on the handset if the call is moved to the handsfree (F0015477)
- When dialing really long number of over 100 characters the MTFIF application crashes (F0015478)
- A local ring-back tone is not played when making an outgoing call for an Avaya ARD VPW (F0015498)
- The dynamic key is left showing a failed call if an Avaya ARD VPW is selected at the same
time as an incoming call to the same VPW is received (F0015499)
- The labels for speaker keys, handset keys and dynamic keys do not match the VPW key (F0015500)
- The speaker keys and handset keys do not show "CONFERENCE x" when a VPW has barged in calls (F0015501)
- When transferring a Cisco call which is also on a speaker channel, if the transfer attempt
is aborted before the second leg is answered by pressing the clear key, the speaker key is
updated with the wrong label (F0015502)
- Incorrect behaviour when attempting to make an Avaya call private if already made
private by the remote party. Both the apperance key and the handset key may be drawn
incorrectly after the privacy attempt fails (F0015503)
- The overall state is incorrect when taking a call off hold with the menu active causing
the menu system to no longer work until OK is pressed (F0015504)
- Various minor CDR bugs (F0015505)
- When barging into a Cisco line the redial menu is updated with the label "Conference"
and attempting to dial out using this always fails (F0015506)
- When barging into an Avaya line if the remote party can be matched in one of the directories
and if the directory entry has a blank long label then the placed call is not recorded
in the call register (F0015511)
- If another user barges into a SIP call, the handset is only updated to display conference
if the appearance is on the displayed page (F0015517)
- Blank long labels are not handled in the call info menu. They are shown as blank lines (F0015518)
- When answering a VPW on a speaker channel, some errors are seen in the messages file. The
call appears to still work OK even with the errors (F0015519)
- Cisco line expiry timer clears the established call after a minute when using redial to make
an outgoing call after it has been seized manually by pressing a line or call appearance (F0015522)
- When an iD808 has no Cisco privacy configured and the user pre-selects privacy and seizes
a Cisco line or call appearance the warning that the privacy request has been cancelled is not
reported (F0015523)
- With a VPW on a speaker channel and the selected handset idle, after making an outgoing call on the
speaker channel pressing the dialpad has no affect. It should initiate manual dialling (F0015526)
- When trying to barge in to a private Cisco call the call attempt fails as expected but the user is
incorrectly presented with "Action not possible" (F0015527)
- The network and icms status icons may not display correctly after a re-power. They are shown
correctly when the menu is displayed but returned to the incorrect colour when the menu is exited (F0015528)
- When the bottom key of screen A is a dynamic key and is showing a held call, changing the privacy
state of Avaya and Cisco calls can result in the dynamic key showing the wrong call state (F0015530)
- When the first call leg of a transferred call is cleared by the far end while the second call
leg is established putting the second call leg on hold does not generate music-on-hold at the far end (F0015531)
- When using a Call Appearance as a VPW for an Avaya PBX, call forwarding should not apply to the VPW
as it is for this user only (F0015532)
- After clearing a conference containing 2 SIP calls with both calls on different speaker channels,
the next SIP call made doesn't work. The call connects correctly but there is no audio in either
direction (F0015534)
- When taking an ARD call off hold by pressing the appearance key while the menu is up, the
call is not cleared from the dynamic key (F0015535)
- ARD calls can ring with the incorrect profile on dynamic keys. This occurred with an ARD on a
non-paginating key with alerting enabled and on a speaker channel and set for a different
alert than page 1 (F0015537)
- If a directory entry doesn't have a long label entered then missed and received calls can
be added to the call register as blank lines (F0015540)
- If a device is set up with no telephony appearances then the SIP icon is red with the error
"9 lines to register - all lines have failed to register." (F0015550)
- If an ARD call is assigned to an empty speaker channel that is part of an active speaker group,
the ARD call appears to stay active but the gooseneck does not work (F0015551)
- When the speaker source is set for handset 1/2 and a group talk key is pressed only one call is
moved to the handset and the others flash indicating a member of the group is on the handset (F0015552)
- When the speaker source is set for handset 1/2 and a group talk is active and there are multiple
active speaker channels. When the handset clear key is pressed the speaker channels that are part
of the active group remain active on the speaker channel with the gooseneck microphone active (F0015553)
Known Defects/Issues in Version 1.200.4.0
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
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New Features/Enhancements Added in Version 1.200.3.0
- Improve bad block detection for ONFI compliant NAND Flash ICs
Defects Resolved in Version 1.200.3.0
- The speaker settings->speaker source menu always lists all options even when handset 2 is not available
- When editing a speaker channel (e.g. priority/latching) the fixed keys xml is sent to i
cms even though nothing has changed
- The temporary *_partial.xml files are not deleted if the update to i cms fails
- When adding/deleting an invalid voice service (voice service with no subnet address configured)
the i cms icon and profile error message are not updated until the device is synchronised
- When answering a call on a speaker channel at the same time as it is cleared from the far end some
internal errors are produced
- When editing a speaker channel the ALR setting can be changed even when the priority is set to 1. When
the priority is 1 the ALR should be forced to Off
- When starting up multiple DSP errors are seen on the internal serial interface for the DSP
- After changing the default line for a directory entry the directory list continues to show the first
entry not the default entry
- Missing warning message "Cannot clear key" when pressing assign-assign followed by an SbRTP call
when the call is assigned to the handset
- When a ringing SIP call is answered on a speaker channel while part of an active group the call
sometimes jumps to the handset muting the microphone for the other members of the group
- Active Cisco calls on speaker channels are not cleared at the far end when a device is synchronised
- A conference on hold is always moved to the selected handset when taking off hold by pressing a
speaker channel. It should stay on the speaker channel. Also the appearances do not show the icon
for the speaker channel the conference is on
- If a conference is on multiple speaker channels latching different members of the conference can
cause the icons to be misleading as they can show mic disabled when the mic is active or mic active
when the mic is disabled
- Appearance lines with blank long labels appear to be listed in a strange order when adding/editing a line
- When an iD808 has an Avaya appearance that is not on a key the device reports a subscription error
- When adding or removing SbRTP channels with VLAN enabled a debug message is displayed in the /var/log/messages file
- If a user is seated on a device and is configured to use 9 Cisco virtual phones and another user
with fewer vitual phones is then seated on the device the SIP icon goes yellow and attempts to
register lines from the old user. Synchronising does not fix the problem but re-powering does
- When in keyfinder and the administrator logs the user out from i cms the keys are all shown in blue
- For incoming Cisco calls that are put on-hold (including via transfer) music-on-hold is not heard
at the far end and he common lamping shows the call as busy-elsewhere instead of on-hold-elsewhere
- When taking an on-hold-elsewhere Cisco call off hold using a dynamic key the call state icon on the
appearance key is left in the connecting call state
- When an iD808 has Cisco lines allocated by i cms but fails to retrieve the master config file
from the Cisco TFTP server the user cannot make or receive Cisco calls (service unavailable) but the
SIP server icon is still green and no error as reported in the SIP server status screen
- DSP diagnostic logging (through DebugDSP) may not work if it has been more than 23 days since the last use
- An Avaya appearance key can be left showing "CONFERENCE 2" when going into and out of privacy (if it is on
a paginating key with an internal index of 0)
- When seizing an Avaya line the error message "**ERROR:vReInviteReceivedCallHelper:
RvSipMsgGetRequestUri failed" is generated
- Error message "**ERROR:UI_RegisterThread: Invalid index (-1) for bulk registration" generated
during logout/resync
- When a conference is a member of an active speaker group and on the handset and cleared off the handset,
remaining members of the speaker group may not be activated depending on the order of the calls in
the speaker group
- CDR heartbeat event does not include speaker of active call
- If the speaker source is set to handset 1 or 2 selecting speaker channels while the speaker settings
menu is displayed does not grey out the speaker source menu option
- Error message "**ERROR:vMtfifSubscribe:appl/mtfif/mtfif_subs.c: RvSipSubsDetachOwner failed (-12)"
generated during logout/resync
- Entering a large voice mail address and then synchronising results in the device losing contact with i cms
- With latching mode set to push to latch the speaker source menu option becomes unavailable
- If an inactive SbRTP channel is added to a key and then deleted it cannot be added a second time
- When in a Cisco call, if a colleague barges into the call the Caller ID correctly changes to "Conference"
but then changes back to the Caller ID of the original far end contact
- With an ARD on a fixed key, if that ARD is replaced by a speaker key using key finder the device still
receives incoming calls for that ARD channel
- Internal errors messages generated when trying to barge in to a busy-elsewhere Cisco appearance with
no remote identity as this call has been made private
- When synchronising, an error can be seen "Invalid speaker item key index"
- Personal directory should be limited to 500 entries (F0012504)
- The Delete option for VPWs in the iD808 UI should not be present as a selectable option. It should
be greyed out (F0015192)
- When a conference is cleared from the far end while on a speaker channel the call failed tone
starts but doesn't stop (F0015196)
- When entering a SIP URL for dialling the string entered can erroneously match a dial plan entry (F0015216)
- When an iD808 is configured with a static IP address and the DNS server address is set to 0.0.0.0 then
the "Show Network" screen reports the last DNS address received from a DHCP server instead
of 0.0.0.0 (F0015226 & F0015227)
- Call forward on busy using a Cisco PBX forwards the call even when the line is not busy (F0015229)
Known Defects/Issues in Version 1.200.3.0
- Screens can get into a strange state after cancelling a call at the same time as the screen saver is about
to be activated (F0013975)
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- On Cisco lines with user A connected to user B and user B performing an unattended transfer to user
C, user A should hear a ringback tone but doesn't (F0015230)
- Ringback tone issue during an attended transfer of a call using the Cisco PBX (F0015228)
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New Features/Enhancements Added in Version 1.200.2.0
- The call state icon for MRD calls has been changed to display busy elsewhere, etc call states instead
of permanantly showing connected
Defects Resolved in Version 1.200.2.0
- When editing/adding an appearance, invalid appearances (e.g. voice services with no address) are still
listed even when the invalid appearance is already on another key
- When using a speed dial to dial out, the handset shows the style of the speed dial and not the appearance
which means that when the call is assigned to a speaker channel it continues to show the speed dial style.
The stye of the speaker channel should always match the appearance style
- Conference containing a SIP call and an ARD call on hold on a dynamic key set for dual line mode shows the
ARD label on both lines. The appearance keys and speaker channels show conference on the second line
- With DHCP status Bad and Ethernet 2 set to loop-through but down the network icon is yellow. It should be red
- Device Ethernet net 2 function and VLAN settings were changed in safemode and the device restarted. When the
device started up again the settings had not been saved
- When VLAN settings are changed from i cms the device is not restarted. It also appears a response to the
profile download is not sent to i cms
- MAC1_ADDRESS CM variable set to blank when changing VLAN settings
- Miscellaneous issues with the displayed caller ID when barging into calls with a Cisco PBX
- Changes made to the Cisco registration to support modified cop files received from Cisco including
blocking incoming and outgoing Cisco calls if the service is unavailable
- When an appearance is moved to or from or between fixed keys the link to any speaker channels in
i cms is lost
- If a call on a speaker channel is added to a conference and another call is also added to the conference
when all the calls are cleared the speaker channel can show an incorrect label
- If a call on a speaker channel is added to a conference and another call is also added to the conference
if the first call is cleared from the far end the icon on the appearance key does not show the conference
icon
- If a call on a speaker channel is added to a conference and another call is also added to the conference if
the first call is cleared from the far end when it is rung the speaker channel shows the alerting style
- An iD808 can run out of call entries and not be able to make any calls if the Cisco lines are not correctly
configured
- Occassional error message generated **ERROR:vUiHandleSipMsg: Received request from MTFIF for unknown
call id message code 4
- Occassionally the SIP status icon can be yellow despite there being no problems with the registration
- The SIP stack can get into a state where it is using the wrong DNS IP address and hence registrations and calls fail
- Potential memory leak with the Cisco privacy functionality
- A Cisco call connected to a speaker channel but currently on a handset can be disconnected by pressing the
privacy key and then immediately pressing the clear key
- Pressing a group talk key containing an idle ARD, an idle SIP and a listen only hoot call changes the
LED to green for the ARD and listen only hoot calls and produces an error
(UI_get_group_talking:../src/UI_states.c: Invalid call data index -1) for the SIP call
- If the VLAN settings are changed in i cms while an iD808 is logged out the update is sent by i cms and
processed by the iD808 but the iD808 does not successfully send a response back to i cms
- When a speaker channel key with a SbRTP channel is edited the speaker channel cannot be selected afterwards
- Default appearance set to Cisco with a message waiting on the Avaya PBX. Select the MSG waiting soft key and
the call to the voice mail server fails as expected but the UI can crash.
- It is possible to replace a speaker key with another speaker key. This should not be allowed
- The Line ID is blank for the outbound dialling if the long label is blank
- In the "Show i cms" status screen profile error that refer to paginating keys list the internal index as
the reference instead of the page and line number of the paginating entry
- When an iD808 has a message waiting from a Cisco PBX and is re-synchronised the message waiting indicator
initially comes on but then goes off
- When a private-on-hold call is taken off hold by pressing the speaker key the call should be moved to
the handset (calls on speaker channels can not be private)
- When Ethernet 2 function is set to loop-through but the port is down and VLAN is enabled the device
cannot get a DHCP IP address or be contacted. Changing the Ethernet 2 function to Off makes the
device work fine.
- When using Cisco and user A calls user B who has been configured to forward to user C, user A see the
remote caller ID as user B instead of user C
- When calling out using Cisco the user sees a PKID number as the caller ID of the far end
- When RTP capabilities listings order has non-standard codecs like 101 Telephone-event listed before
the standard RTP codecs like G711, G729 and G722, there is no voice
- When a SIP NOTIFY is received for a Cisco line that is in a conference a strange Caller ID is
displayed instead of "Conference"
- Cisco privacy operation is blocked if Avaya privacy is not configured on the unit
- When barging into a Cisco line appearance the Caller ID label on the appearance is blank until the
page is refreshed
- Incorrect handling of Cisco SIP NOTIFY messages can result in an appearance key showing idle when
in a conference call
- Incoming calls from a Cisco PBX are recorded in the call register 3 times (if 2 privacy appearances are configured)
- When upgrading with VLAN enabled the device can announce to i cms with a MAC of 0
- Miscellaneous minor CDR defects
- If a listen only call is assigned to a handset with privacy already requested then the padlock cannot be
cleared from the call while it is on the handset
- Selecting a group talk key for a speaker with a private elsewhere MRD call attempts to talk on the
call (call shows connecting)
- LEDs for appearances on speaker channels is inconsistent. Idle SIP appearance on a speaker channel
is off but idle ARD appearance on speaker channel is green (it should be off)
- Private ARD on hold on a speaker channel removes privacy when the speaker channel is selected. SIP calls get assigned
to the handset when on hold private. ARD calls should behave the same
- Some members of a conference cannot be assigned to the handset from a speaker channel if the
conference is on multiple speaker channels. Only one speaker channel can be assigned
- There is an issue when a conference is on multiple speaker channels and the call(s) are cleared from
the far end. When all the calls are cleared the speaker channels can show incorrect labels and icons
- SbRTP calls cannot be received when VLAN is enabled
- When a user barges into a Cisco conference and then leaves the conference the Caller ID for the
conference is shown as blank and the user cannot rejoin the conference by barging in again
- For Cisco the number of participants listed in the conference label is inaccurate if the device does
not have privacy configured and hence the conference label should be shown as just "Conference"
instead of "Conference x"
- The echo canceller for the gooseneck does not work correctly
- With Cisco lines hold elsewhere calls put on-hold by an iD808 are shown as busy elsewhere
- When an iD808 puts a Cisco call on-hold the far end does not hear any music on hold
- When a Cisco call is put on-hold by a Cisco phone and then taken off hold by an iD808 the iD808
barges into the call and the Cisco phone is left with a call leg to the newly created conference
that is still on-hold
- When making an outgoing Avaya call using an appearance that is also on a speaker channel the
Caller ID is not updated on the speaker key
- Should not allow the user to enter the speaker source menu when any speaker channel are talking and
should block and speaker key presses while in the speaker souce menu with "Action not possible"
- The user is not always blocked from making a Cisco conference call private
- DTMF tones using RFC2833 telephone events are not transmitted correctly when VLAN is enabled
- When transferring a call using Cisco the far end user of the first leg of the call does not
hear music-on-hold
- Caller ID issue: When making a call using a line appearance at user A, user B receives the master appearance
name until the call is answered then the line appearance name is shown (F0012524 & F0014953)
- Caller ID issue when a call received from a number withheld source is transferred (F0014991)
- An iD808 can show more lines subscribed than configured (F0014361)
- Call forwarding does not work properly if you re-save a call forward change too quickly (F0015068)
- Speaker source set to Handset and Push-to-latch creates open handset mic when ARD initiated but should
be set to mute until speaker channel button pressed (F0015071)
- Personal Phonebook adds a '9' as a prefix to a URL address, which is not a phone number (F0015075)
- Unseat a user on a turret configured with a static ip address and then change the IP address and
reseat the user, when the user is back on the turret, all the lines have failed to register and
subscribe. (F0015088)
- The maximum length of the ping address for the ping menu is too short (F0015130)
- Minor correction required for the ping screen help text (F0015135)
- CM variable DNS_HOST_NAME is cleared when the DHCP lease times out and a NETupdate is performed causing
i cms to go out of sync (F0015153)
Known Defects/Issues in Version 1.200.2.0
- The speaker settings->speaker source menu always lists all options even when handset 2 is not available
- When editing a speaker channel (e.g. priority/latching) the fixed keys xml is sent to i
cms even though nothing has changed
- The temporary *_partial.xml files are not deleted if the update to i cms fails
- When adding/deleting an invalid voice service (voice service with no subnet address configured)
the i cms icon and profile error message are not updated until the device is synchronised
- When answering a call on a speaker channel at the same time as it is cleared from the far end some
internal errors are produced
- When editing a speaker channel the ALR setting can be changed even when the priority is set to 1. When
the priority is 1 the ALR should be forced to Off
- When starting up multiple DSP errors are seen on the internal serial interface for the DSP
- After changing the default line for a directory entry the directory list continues to show the first
entry not the default entry
- Missing warning message "Cannot clear key" when pressing assign-assign followed by an SbRTP call
when the call is assigned to the handset
- When a ringing SIP call is answered on a speaker channel while part of an active group the call
sometimes jumps to the handset muting the microphone for the other members of the group
- Active Cisco calls on speaker channels are not cleared at the far end when a device is synchronised
- A conference on hold is always moved to the selected handset when taking off hold by pressing a
speaker channel. It should stay on the speaker channel. Also the appearances do not show the icon
for the speaker channel the conference is on
- If a conference is on multiple speaker channels latching different members of the conference can
cause the icons to be misleading as they can show mic disabled when the mic is active or mic active
when the mic is disabled
- Appearance lines with blank long labels appear to be listed in a strange order when adding/editing a line
- When an iD808 has an Avaya appearance that is not on a key the device reports a subscription error
- When adding or removing SbRTP channels with VLAN enabled a debug message is displayed in the /var/log/messages file
- If a user is seated on a device and is configured to use 9 Cisco virtual phones and another user
with fewer vitual phones is then seated on the device the SIP icon goes yellow and attempts to
register lines from the old user. Synchronising does not fix the problem but re-powering does
- When in keyfinder and the administrator logs the user out from i cms the keys are all shown in blue
- For incoming Cisco calls that are put on-hold (including via transfer) music-on-hold is not heard
at the far end and he common lamping shows the call as busy-elsewhere instead of on-hold-elsewhere
- When taking an on-hold-elsewhere Cisco call off hold using a dynamic key the call state icon on the
appearance key is left in the connecting call state
- Personal directory should be limited to 500 entries (F0012504)
- Screens can get into a strange state after cancelling a call at the same time as the screen saver is about
to be activated (F0013975)
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
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New Features/Enhancements Added in Version 1.200.1.0
- Support for enhanced telephony features with a Cisco PBX (common lamping, barge-in, privacy)
- Persistence for speaker channels introduced (replaces all previous speaker channel types)
- Speaker pages introduced including the "Speaker Page" menu option
- Speaker channel priority and automatic level reduction introduced
- Support for operation of the speaker channels without a gooseneck microphone added
- Configurable latching mode for speaker channels (tap-latching or push-to-latch)
- Group talk key added as a new key type. The group talk keys are programmable via keyfinder and the groups
are programmable via the Program->Special Keys->Program Group menu option. The group talk key operation
also includes the line to handset while group broadcasting feature
- Programmable voice activity indicator period for speaker channels
- Spanning Tree Protocol supported to provide redundancy for the Ethernet interface
- VLAN supported
- Enhanced diagnostics support via Engineering Tools->Log Settings and Engineering Tools->Send Logs
- Leading zero removed for the hour displayed on the date and time footer bar
- A secondary DNS server and NTP server supported
- Network status subdivided into Network, i cms and SIP server status screens and the SIP server status
enhanced to show all lines registered and subscribed as well as those that have failed
- SSH sessions automatically time out after 20 minutes of inactivity
- Alert override status icon added to the status bar
- Support for orphaned calls (when the associated appearance is deleted while the call is still active)
- Call forward soft key changed to a shortcut to menu key
- Sidetone now played on both handsets and headsets but only if the handset or headset is being used
for an active call and when the handset or headset is not muted
- Auto select idle handset configuration option added
- Long labels are now optional. When a long label would be displayed but is blank the short label will
be displayed instead
- Clear speaker menu option changed to "Unassign Speaker". Assign-Assign key press is still used to
clear the calls off a speaker channel
- "Speaker Settings" menu added
- CDR enhanced to support new functionality and to generate events when the received audio on
active speaker channels is detected or lost
- Support added for sending DTMF tones using RFC2833 Telephone Events for telephony calls
Defects Resolved in Version 1.200.1.0
- Receive audio not decoded with some third party phones (where the audio packet size differ from
the expected size) (F0011734, F0012090, F0013583)
- The network receive fifo does not reset if an overflow is detected (caused by corrupted network
packets being received) (F0012009)
- Occassionally there is a yellow network icon after re-synchronise with the network status showing
an error message of "restart occurred" under the DSP Status (F0012536)
- When receiving an incoming call from a device that has dialled a direct sip URL the iD808 rings on
the first appearance in TFTP screen order instead of the first appearance in appearance number
order (F0012552)
- When using the G711 codec if the network re-orders packets then some audible artifacts are heard (F0012735)
- RTCP Receiver reports not analysed correctly (F0013145)
- Occasionally no audio between a Snom phone and an iD808 using Cisco G722 (F0013170)
- Squeak heard when taking a Cisco call off hold (F0013171)
- If RTCP is enabled RTCP packets can be gererated with a MAC address of zero (F0013453)
- Clicking sound heard after answering a G711 call with a Snom phone on handsfree and then picking up
the handset (F0013601)
- A key was left visible on a display when the device was synchronised (F0013615)
- There can be muting problems if a listen only hoot channel is moved on to a handset (F0013680, F0014131,
F0014599, F0014649)
- In the DebugDsp command interface the CONFIGURE command reports the IP addresses that have been
registered by the PowerPC. These quad dotted IP addresses are always reported with the third field
as 0 regardless of their actual value (F0013888)
- The DSP does not pass through Multicast UDP packets with odd port numbers to the PowerPC even when
the PowerPC has registered the IP address with the DSP (F0013889)
- The iD808 may not announce to i cms after being manually reconfigured in safemode (F0013896)
- The iD808 UI allows the user to add an entry with only a space resulting in what appears to be a
blank speed dial key on the iD808 (F0013968)
- Removing a mixer from the DSP code without first removing all of its inputs and outputs may produce
unpredictable results (F0014025)
- When using Cisco Version 7.1.2.1.10000-16 when you place a call on HOLD and repeatedly take on and
off HOLD rapidly the call will drop (F0014056)
- When editing a speaker key on an iD808 using keyfinder the user is presented with "PROFILE"
& "PRESENCE" entries. They can never be selected but they should not have been shown as these will
be inherited from the appearance key that is assigned to the speaker key and hence are not
properties of the speaker key (F0014090)
- Very occasionally the bottom key display area of screen A is cached as a blank key instead of its
real display type (F0014105)
- When showing a ringing call on a speaker, the alert profile always uses the default alert
profile and not the profile of the call assigned to the speaker key (F0014145)
- When assigning a call from a speaker channel to the handset and selecting the other handset,
the call cannot be cleared from the first handset if the second handset is idle (F0014154)
- If a SIP call in the "far end ringing" state is on a handset and a speaker channel the ringback tone
can be heard on the handset. If the CLEAR button is pressed, the call clears off the handset but
the tone remains (F0014156)
- It should be possible to send the iD808 RTP recording streams to either a Unicast or Multicast IP
address. i cms will allow you to configure either type of IP Address. However when the iD808
recording streams are configured to send RTP traffic to a multicast address no RTP audio traffic
is sent (F0014157)
- When receiving an incoming SIP that contaions CLI with non-UTF-8 characters the UI crashes. This
happens with an Avaya Integral Enterprise 55 PBX when receiving a German Umlaut 'ä' (F0014205)
- Occasionally, the SIP registration timeout for some lines can be stuck at 4 seconds (F0014211)
- The overall state is not updated when selecting a speaker channel containing a SIP call on hold. The
call is moved to the selected handset but as the overall state of the device is not updated, DTMF
tones do not work and other functionality is unpredictable (F0014315)
- When no default appearance is present on an iD808 and a voice mail message is present, the LED and
system soft key appear but the soft key cannot be selected as there are no lines available to dial
out on. No error message is given when the button is pressed (F0014969)
Known Defects/Issues in Version 1.200.1.0
- The speaker settings->speaker source menu always lists all options even when handset 2 is not available
- When editing a speaker channel (e.g. priority/latching) the fixed keys xml is sent to i
cms even though nothing has changed
- The temporary *_partial.xml files are not deleted if the update to i cms fails
- When editing/adding an appearance, invalid appearances (e.g. voice services with no address) are still
listed even when the invalid appearance is already on another key
- When adding/deleting an invalid voice service (voice service with no subnet address configured)
the i cms icon and profile error message are not updated until the device is synchronised
- Miscellaneous minor CDR defects
- When answering a call on a speaker channel at the same time as it is cleared from the far end some
internal errors are produced
- When editing a speaker channel the ALR setting can be changed even when the priority is set to 1. When
the priority is 1 the ALR should be forced to Off
- When starting up multiple DSP errors are seen on the internal serial interface for the DSP
- After changing the default line for a directory entry the directory list continues to show the first
entry not the default entry
- When using a speed dial to dial out, the handset shows the style of the speed dial and not the appearance
which means that when the call is assigned to a speaker channel it continues to show the speed dial style.
The stye of the speaker channel should always match the appearance style
- Conference containing a SIP call and an ARD call on hold on a dynamic key set for dual line mode shows the
ARD label on both lines. The appearance keys and speaker channels show conference on the second line
- With DHCP status Bad and Ethernet 2 set to loop-through but down the network icon is yellow. It should be red
- Device Ethernet net 2 function and VLAN settings were changed in safemode and the device restarted. When the
device started up again the settings had not been saved
- When VLAN settings are changed from i cms the device is not restarted. It also appears a response to the
profile download is not sent to i cms
- MAC1_ADDRESS CM variable set to blank when changing VLAN settings This may have occurred because the VLAN
settings were changed from i cms and the device was not restarted
- Miscellaneous issues with the displayed caller ID when barging into calls with a Cisco PBX
- Changes are required to the Cisco registration to support modified cop files expected shortly from Cisco
- When an appearance is moved to or from or between fixed keys the link to any speaker channels in
i cms is lost
- If a call on a speaker channel is added to a conference and another call is also added to the conference
when all the calls are cleared the speaker channel can show an incorrect label
- If a call on a speaker channel is added to a conference and another call is also added to the conference
if the first call is cleared from the far end the icon on the appearance key does not show the conference
icon
- If a call on a speaker channel is added to a conference and another call is also added to the conference if
the first call is cleared from the far end when it is rung the speaker channel shows the alerting style
- Personal directory should be limited to 500 entries (F0012504)
- Caller ID issue: When making a call using a line appearance at user A, user B receives the master appearance
name until the call is answered then the line appearance name is shown (F0012524 & F0014953)
- Screens can get into a strange state after cancelling a call at the same time as the screen saver is about
to be activated (F0013975)
- There is a problem with transfer when interoperating with a Mitel PBX (F0014210)
- An iD808 can show more lines subscribed than configured (F0014361)
- Caller ID issue when a call received from a number withheld source is transferred (F0014991)
- Issue with the Avaya FNE (Feature Name Extension) 'Transfer on Hangup - 2971' (F0014997)
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New Features/Enhancements Added in Version 1.130.1.0
- Early media support added to SIP telephony calls
Defects Resolved in Version 1.130.1.0
- No defects resolved in this release
Known Defects/Issues in Version 1.130.1.0
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New Features/Enhancements Added in Version 1.120.2.0
- No features or enhancements added to this release
Defects Resolved in Version 1.120.2.0
- An unattended transfer sometimes fails as a result of sending a BYE too early (F0015390)
- With 200 ARDs and a single SIP line the SIP status icon is red after a synchronise
and only goes green after the next re-registration attempt (F0015391)
- An unattended transfer sometimes fails as a result of the SIP stack responding slowly (F0015400)
Known Defects/Issues in Version 1.120.2.0
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New Features/Enhancements Added in Version 1.120.1.0
- Maximum number of SbRTP calls increased from 60 to 200 (CN2682)
- Low bandwidth SbRTP mode introduced. Configurable via i cms (CN2682)
- CM variables added to monitor last read time of various message queues for
diagnostics purposes
Defects Resolved in Version 1.120.1.0
- Updating of keys when in key finder is really slow when multiple MRD common lamping changes occur at
the same time (F0015277)
- Memory leak if the UI to MTFIF registration message is full and a registration message is
discarded (F0015278)
Known Defects/Issues in Version 1.120.1.0
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New Features/Enhancements Added in Version 1.110.16.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.16.0
- The iD808 devices can suffer a UI lockup and fail to synchronise and reboot successfully
when placed within a subnet that has high SbRTP traffic (F0015266)
Known Defects/Issues in Version 1.110.16.0
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New Features/Enhancements Added in Version 1.110.15.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.15.0
- A possible but rare UI crash when moving an active ARD or MRD call to a handset by pressing the appearance key (F0014951)
Known Defects/Issues in Version 1.110.15.0
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New Features/Enhancements Added in Version 1.110.14.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.14.0
- Speed dial etc on an Avaya PBX can occasionally fail to connect because the line seize call is
cleared down too early. This has been observed in a customer configuration with an appearance
bridged to an analogue phone (F0014947)
Known Defects/Issues in Version 1.110.14.0
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New Features/Enhancements Added in Version 1.110.13.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.13.0
- When dialling out using a speed dial, directory entry or call register entry the destination
number shown on the appearance and handset key is replaced by the number of the appearance
being used to make the outgoing call. This defect was introduced by the changes made to 1.110.13.0
(F0014940)
Known Defects/Issues in Version 1.110.13.0
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New Features/Enhancements Added in Version 1.110.12.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.12.0
- An external call that has been answered by a colleague and then barged in to remains
with a caller ID of "CONFERENCE" after the colleague drops out of the call and
hence it is not possible to transfer the call or make it private. This is a
different viariation of the problem report F0014925 listed below (F0014934)
Known Defects/Issues in Version 1.110.12.0
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New Features/Enhancements Added in Version 1.110.11.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.11.0
- An external call that has been answered by a colleague and then barged in to remains
with a caller ID of "CONFERENCE" after the colleague drops out of the call and
hence it is not possible to transfer the call or make it private (F0014925)
Known Defects/Issues in Version 1.110.11.0
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New Features/Enhancements Added in Version 1.110.10.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.10.0
- Possible but rare UI crash during resynchronisation (F0014914)
- Dynamic keys may not be updated correctly if a new NOTIFY message or re-INVITE is received
for an existing ringing incoming call (F0014912)
Known Defects/Issues in Version 1.110.10.0
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New Features/Enhancements Added in Version 1.110.9.0
- SIP registration enhanced for the generic PBX type to add the port parameter to the
contact header to allow successful registration to a Mitel PBX
Defects Resolved in Version 1.110.9.0
- Dynamic keys may not be updated correctly if a NOTIFY messages is received for an incoming
call or for a call on hold elsewhere if there is no associated appearance key configured
on the iD808 (F0014904)
Known Defects/Issues in Version 1.110.9.0
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New Features/Enhancements Added in Version 1.110.8.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.8.0
- Incorrect warning message displayed when the transfer key is pressed while a conference
is active on the selected handset (F0014887)
- When a user has a SIP call on a dynamic speaker channel and on a handset and if another user
barges in (including activating privacy) the handset text is updated to show CONFERENCE
x indicating the number of people in the call but the dynamic speaker channel is not
updated correctly resulting in displaying different information (F0014896)
- When a user has a SIP call on a dynamic speaker channel and on a handset and tries to
activate privacy, if the privacy attempt fails, possibly because the call is to a local
user who has already made the call private, the handset correctly changes to a grey padlock
which then gets removed but the speaker channel shows the grey padlock but then the grey
padlock does not get removed (F0014897)
- When receiving an unattended transfer call it is possible for the same call to be
simultaneously shown on two dynamic keys. Only one of these keys can be used to successfully
answer the call (F0014898)
Known Defects/Issues in Version 1.110.8.0
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New Features/Enhancements Added in Version 1.110.7.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.7.0
- On an Avaya PBX when User A calls User B whose phone is configured to forward to
User C, User A sees User B as the caller ID instead of User C. This bug was
introduced in the previous release, v1.110.6.0 (F0014879)
Known Defects/Issues in Version 1.110.7.0
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New Features/Enhancements Added in Version 1.110.6.0
- Enhancements to display "CONFERENCE n" when users barge in to the local end
of a call on an Avaya PBX
- Enhancements to allow all users, whether an originator, answerer or barged-in
user, to make a call private or transfer a call but only if they are the only
local user for the call i.e. there are no other barged-in users on the local
end of the call
- U-boot support for new NAND flash device (Micron MT29F2G08AADWP/D) added
- Upgrade process enhanced to block upgraders that may stop the device from
operating correctly
Defects Resolved in Version 1.110.6.0
- Occassional an outgoing call attempt on an Avaya PBX fails to ring at the
far end and the user has to clear the call and dial again (F0014746)
- When the dialpad is in alphanumeric mode and the user is cycling through
miscellaneous characters using the "1" key some 'symbols' are incorrectly
displayed (F0014790)
Known Defects/Issues in Version 1.110.6.0
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New Features/Enhancements Added in Version 1.110.5.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.5.0
- It is possible for all SIP communications to stop when repeatedly trying to register a telephony
line that is rejected by the PBX requiring the device to be re-powered to rectify the problem (F0014644)
Known Defects/Issues in Version 1.110.5.0
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New Features/Enhancements Added in Version 1.110.4.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.4.0
- When transferring a call, two dynamic keys can show the same call on-hold if the second leg is
cleared from the far end. One dynamic key shows the call as on-hold-elsewhere and the other shows
the call as on-hold-here (F0014580)
- Answering a VPW does not clear the call off of a dynamic key (F0014586)
Known Defects/Issues in Version 1.110.4.0
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New Features/Enhancements Added in Version 1.110.3.0
- The operation of the dynamic keys has been enhanced. Each dynamic key can now
be configured to only display on-hold and on-hold elsewhere calls, only display
ringing calls or display both on-hold and ringing calls
Defects Resolved in Version 1.110.3.0
- No defects resolved in this release
Known Defects/Issues in Version 1.110.3.0
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New Features/Enhancements Added in Version 1.110.2.0
- No features or enhancements added to this release
Defects Resolved in Version 1.110.2.0
- SIP NOTIFY messages received for appearances that are not allocated to any keys on the
device can still result in an incoming call or an on-hold call being shown on a dynamic key
(F0014492)
Known Defects/Issues in Version 1.110.2.0
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New Features/Enhancements Added in Version 1.110.1.0
- The operation of the dynamic keys has been enhanced. The dynamic keys can now
display on-hold and on-hold-elsewhere calls as a configuration option and these calls
can be selected on the dynamic key to take them off-hold. Also dynamic keys can
now be configured to display all relevant calls or only those whose appearances are
currently hidden. These two features are configurable via i manager.
Defects Resolved in Version 1.110.1.0
- Compatibility issues with Avaya CM version 5.2 that can result in appearances appearing
busy when they should be idle and problems with configuring call forwarding (F0014288,
F0014289, F0014290, F0014366)
- It is possible during PBX fault conditions for the SIP stack to run out of resources
making it impossible to make an outgoing call after the PBX fault condition is resolved
(F0014259)
Known Defects/Issues in Version 1.110.1.0
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New Features/Enhancements Added in Version 1.102.1.0
- U-boot support for new NAND flash device (Micron MT29F2G08AADWP/D) added
- Upgrade process enhanced to block upgraders that may stop the device from
operating correctly
Defects Resolved in Version 1.102.1.0
- No defects resolved in this release
Known Defects/Issues in Version 1.102.1.0
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New Features/Enhancements Added in Version 1.101.1.0
- No features or enhancements added to this release
Defects Resolved in Version 1.101.1.0
- The hands-free audio can be transmitted at a volume level that distorts on some
networks (F0014096)
Known Defects/Issues in Version 1.101.1.0
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New Features/Enhancements Added in Version 1.100.16.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.16.0
- The call register can fail to initialise correctly and then will not be available
to the user (F0014016)
Known Defects/Issues in Version 1.100.16.0
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New Features/Enhancements Added in Version 1.100.15.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.15.0
- The UI can crash when receiving a call when an error has previously occurred when
loading the directory (F0013993)
- The MRD ring signal can continue to be sent after releasing the key by having the
soft key on a paginating key and changing page while the key is still held down (F0014015)
- Cannot make outgoing calls when using Cisco Call Manager 7 (F0014047)
- Soft keys are not updated when answering a call with the UI menu active (F0014062)
- When pressing a page navigation key, followed by the OK key, followed by the back key very
quickly the device can get into a state where auto page repeat is stuck on causing the device
to continue paging until a navigation key is pressed (F0014069)
Known Defects/Issues in Version 1.100.15.0
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New Features/Enhancements Added in Version 1.100.14.0
- The on-hold elsewhere illuminated key indicator changed from a solid amber to
a slow flashing amber indication (CN2634)
- The call failed timeout period reduced from 5 seconds to 1.2 seconds (CN2634)
Defects Resolved in Version 1.100.14.0
- The voice activity indicator for a MRD call on a speaker channel incorrectly overrides any
alerting indicator being shown on the illuminated key (F0013988)
Known Defects/Issues in Version 1.100.14.0
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New Features/Enhancements Added in Version 1.100.13.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.13.0
- The volume softkey does nothing when pressed with a listen only hoot channel on the
handset (F0013880)
- When dialling from a directory entry if an appearance is seized prior to dialling the
dial number is not converted using the prefix dialling rules (F0013897)
Known Defects/Issues in Version 1.100.13.0
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New Features/Enhancements Added in Version 1.100.12.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.12.0
- Sometimes changes to the Device IP Address made in safe mode are not saved (F0013786)
- Sometimes a busy-elsewhere ARD line is incorrectly shown as idle after a
re-sync from i cms (F0013815)
Known Defects/Issues in Version 1.100.12.0
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New Features/Enhancements Added in Version 1.100.11.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.11.0
- A VPW call using an Avaya call appearance that has been made private cannot be made
un-private (F0013808)
Known Defects/Issues in Version 1.100.11.0
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New Features/Enhancements Added in Version 1.100.10.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.10.0
- When using Asterisk with Call Forward set to Call Forward No Answer and selecting a
different page so that an incoming call will ring on a dynamic key the incoming call
is forwarded after the set time, however the dynamic key is left ringing (F0013777)
- When creating an anonymous call appearance from the iD808 it displays a line appearance
icon instead of a call appearance icon (F0013787)
- VPWs on Avaya using bridged call or line appearances cannot be made private if they
have been originated from the far end (F0013793)
- Under the Device Ethernet configuration of the iD808 the NET Speed is labelled as
Mhz; this should be Mbps (F0013795)
Known Defects/Issues in Version 1.100.10.0
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New Features/Enhancements Added in Version 1.100.9.0
- Full support for ARD dialling using DTMF tones (CN2626)
Defects Resolved in Version 1.100.9.0
- When using the Avaya PBX and manually dialling without seizing an appearance first, the
iD808 automatically chooses the first available appearance. If the line seize attempt of
this appearance fails the iD808 should automatically try the next available appearance
until either an appearance has been successfully seized or else all available appearances
have been tried and have failed. However if at the same time the user starts manually
dialling the chosen appearance is seized from another unit that shares that appearance
the manual dialling attempt can on occassions immediately fail with the user warning that
the line seize attempt has timed out despite not trying the other appearances (F0013526)
- User A has an ARD appearance and make a call which is answered at the far end.
User A now makes that call into a conference and assigns the conference to a dynamic
speaker. There is now only 1 way voice between the 2 units (no audio from User A) (F0013578)
- In the i cms server config edit screen an IP address can be made invalid and then
greyed out by setting "Use DNS" to on. In this situation the save button remains greyed
out instead of being enabled (F0013614)
- Occassional UI crash associated with using VPWs (F0013647 & F0013648)
- Call info screen incorrect with a VPW call (F0013659)
- If an iD808 user makes a change in the "Configure Network" menu, save this change and
then log-out of the device again via the iD808 menu the device shows the message "activating
a configuration change, please wait" and then incorrectly auto-logs back in (F0013661)
- If a VPW is put into a conference and the conference is put on hold then the conference
cannot be taken off hold by selected the VPW appearance (it can only be taken off hold
by pressing the conference button). Also the conference cannot be moved between handsets
or from a speaker channel to the handset by pressing the VPW appearance (F0013663)
- No audio heard after taking a call off hold if the call was put on hold while the call
was on handsfree (F0013667)
- The alert being played is not checked when changing pages when ring displayed page is
enabled. This means that an appearance does not start audibly ringing when the page is
changed to the page the appearance is on. Also, appearances on float keys do not audibly
ring when on a page with ring displayed page enabled (F0013678)
- Missing CDR event when answering an incoming VPW when using asterisk. This only occurs
when selecting the VPW key not when answering the VPW on a dynamic key (F0013679)
- After a resynchronise or power cycle cached versions of paginating keys are not used
until the user changes page or sub-page and hence exiting the menu or screen saver is
untidy (F0013690)
- If a user has permissions to use more than 60 SbRTP channels but less than 60 are assigned
to keys then the i cms status icon incorrectly goes yellow producing the error "Too many
channels". This should not be shown as an error and not turn the i cms status icon
yellow. The i cms status icon should still go yellow when more than 60 SbRTP
channels are assigned to keys (F0013696)
- The audio device associated with a speaker channel is incorrectly changed when moving
a speaker channel. This causes problems when selecting the speaker channel (F0013698)
- Privacy cannot be selected for barged in ARD or MRD calls. This was only introduced in
1.100.8.0 (F0013708)
- When a conference is created with an ARD call and a SIP call and the ARD call is cleared
from the far end the SIP call is left with one way voice. The transmit stream is disabled
until the call is muted and un-muted (F0013723)
- An extra CDR event CONFERENCE_UNHOLD is generated when changing the handsfree setting
for a conference (F0013724)
- The requesting privacy padlock disappears on an idle handset when a warning dialog box
is shown. The requesting privacy status is still on the handset but is not visible (F0013725)
- When a conference call is on the handset and is muted and then is made hands-free it is
incorrectly still muted while the indicator shows it as not muted. A similar scenario happens
when the conference call is moved back from the speaker to the handset (F0013728)
- Editing voice services or appearances on float keys deletes the key (F0013731)
Known Defects/Issues in Version 1.100.9.0
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New Features/Enhancements Added in Version 1.100.8.0
- The user can manually send DTMF tones on a connected ARD line by pressing the dialpad keys
while the ARD call is on the selected handset (part of CN2626)
Defects Resolved in Version 1.100.8.0
- For VPWs information shown on the handset keys changes depending upon whether or not entries
within the corporate directory are ticked as hidden (F0013508)
- The call type is listed as "SIP Outbound" in the "Info" screen when answering a line appearance
or bridged call appearance on an Avaya PBX (F0013540)
- A device with no expansion modules can incorrectly state that there are not enough float
keys (F0013543)
- When playing alerts, showing alerting on an appearance key or populating a dynamic key the
alert profile to reference should be the one against the key when the alert setting for the
key is on or the default alert profile when the alert setting for the key is off. Sometimes
the alert profile against the key is used instead of the default alert profile and this can
result in the wrong ringtone being played or the wrong style being used or the dynamic keys
being populated in the wrong order (F0013544)
- Changing the volume using the volume softkey but leaving the volume at the same level sends
out an update to i cms which does not contain an update to any variables. This update
does not cause any problems for i cms or the iD808 but does not need to be sent as no
variables have been changed (F0013549)
- Silence suppression does not work for G.729 or G.711 (F0013552)
- When transferring a call on an iD808 if the user has not pressed any keys for the last 15
minutes the transfer attempt should be aborted and the user should be returned to the original
call but this never happens (F0013559)
- With an active SIP call on the selected handset if the user presses a dialpad key and then
before releasing it assigns the call to a dynamic speaker channel the DTMF tone continues
to play. It should have been cancelled when the call was assigned to the dynamic speaker
channel (F0013581)
- Key page variables are not correctly updated when moving keys from non-page keys to page keys
or vise versa. This can cause some pages to be skipped over as they are believed to be empty.
When synchronised the device shows some errors but recovers and the missing page is not
skipped over (F0013589)
- During startup there are many kernel messages logged to the messages log file as warnings
or errors that should have been logged as informational or notice level (F0013591)
Known Defects/Issues in Version 1.100.8.0
- When using the Avaya PBX and manually dialling without seizing an appearance first, the
iD808 automatically chooses the first available appearance. If the line seize attempt of
this appearance fails the iD808 should automatically try the next available appearance
until either an appearance has been successfully seized or else all available appearances
have been tried and have failed. However if at the same time the user starts manually
dialling the chosen appearance is seized from another unit that shares that appearance
the manual dialling attempt can on occassions immediately fail with the user warning that
the line seize attempt has timed out despite not trying the other appearances (F0013526)
- User A has an ARD appearance and make a call which is answered at the far end.
User A now makes that call into a conference and assigns the conference to a dynamic
speaker. There is now only 1 way voice between the 2 units (no audio from User A) (F0013578)
- Support for ARD dialling using DTMF tones not fully implemented (CN2626)
New Features/Enhancements Added in Version 1.100.7.0
- checksipdomain utility added to filesystem
- cap_dns.sh script added to filesystem
Defects Resolved in Version 1.100.7.0
- With auto handset mute set to ON and a single SIP call in progress, if you move the call
from handset 1 to handset 2 then handset 1 becomes muted (F0013364)
- The call forward save button is greyed out when accessing via the soft key. This only
occurs when cancelling out of the add speed dial menu (F0013454)
- Cancelling the second leg of an attended transfer clears the entire call instead of going
back to the first leg of the call (F0013455)
- Soft keys are not updated when taking a SIP call off hold by pressing the dynamic speaker
channel (F0013462)
- Local muting does not work when selecting speaker channels with conferences on them (F0013463)
- Hoot and MRD speaker channels can not be selected when attempting to clear a conference
containing these calls (F0013464)
- An extra CDR event "TALK_STATE_CHANGED" is generated when pressing or releasing an ARD speaker
channel in the connecting state (F0013465)
- Listen only status for hoot appearances is not copied when moving between handsets which can
cause listen only channels to get talk rights when moving to a speaker channel (F0013466)
- Minor help text corrections (F0013481)
- An erroneous error is logged when selecting an unassigned paginating key when in delete
float key finder mode (F0013482)
- The latching setting is not remembered when moving speaker channels so all speaker channels
moved end up with latching off (F0013483)
Known Defects/Issues in Version 1.100.7.0
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New Features/Enhancements Added in Version 1.100.6.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.6.0
- The soft keys refresh after using double assign to clear a voice service (F0013275)
- When insufficient float keys are present on an iD808 the i cms icon is yellow. If extra
float keys are then added from the iD808 UI to clear the error the i cms icon stays
yellow until the unit is re-synced. This problem was partially fixed in 1.100.5.0 but
there were some outstanding aspects that have now been fixed by this release (F0013338)
- An empty float key has the style of the key which should be floated to it on the iD808
layout page when viewing the fully expanded iD808 (when all float keys are empty) (F0013371)
- Audio can be lost after making a call private and then after a while the call clears down
at one end (F0013375)
- In the login screen if you type a letter in the password box and then immediately press #
the letter you just typed does not get hashed out until the next key entry (F0013386)
- Paginating soft keys are not updated when moving from a page containing soft keys in the
same position (F0013391)
- A bridge call appearance icon is shown as a call appearance icon when selecting and clearing
it when on a float key (F0013392)
- After seating a different user to the one previously seated on an iD808 the device can
report an "appearance entry not available" profile error (F0013395)
- Entering a long page title in i cms using characters that are multi-byte when coded
using UTF8 can cause the iD808 UI to crash on a resync (F0013419,F0010350)
- An iD808 occassionally may have all 3 icons flashing red after a power cycle despite reporting
no errors in the Network status (F0013429)
- When a SIP server fails to respond to a SIP response packet sent by iD808 and timed out after
(approx. 30 seconds) when call is already established the iD808 sends the SIP Bye to clear
the remote end but does not clear its own side of call (F0013436)
Known Defects/Issues in Version 1.100.6.0
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New Features/Enhancements Added in Version 1.100.5.0
- No features or enhancements added to this release
Defects Resolved in Version 1.100.5.0
- With a large network latency and dialling a number that also rings on the calling iD808
the call state icon changes from connecting to established even though the call has not
been answered (F0013266)
- When editing a speed dial or anything else that requires a cached key to be updated there
can be a noticeable period (several seconds) where the UI freezes (F0013299)
- At the login screen, in the password box if you type a letter then immediately press # to
change to numbers you lose the flashing cursor and have to press ok again to reselect
the password box (F0013351)
- The outbound dial string is not cleared when synchronising the device so if a device is in
the outbound dialling screen when synchronised then numbers entered after the device is
logged on will be added to the dial string (F0013352)
- Entering the # when in the password entry box causes problems with the password (F0013359)
- Ringing bridged call appearances can be added to dynamic keys on non-fitted expansion units
and hence will not be visible on the unit (F0013369)
Known Defects/Issues in Version 1.100.5.0
- When insufficient float keys are present on an iD808 the i cms icon is yellow. If extra
float keys are then added from the iD808 UI to clear the error the i cms icon stays
yellow until the unit is re-synced. This problem has been partially fixed in 1.100.5.0 but
their are some aspects that are still outstanding (F0013338)
- Audio can be lost after making a call private and then after a while the call clears down
at one end (F0013375)
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New Features/Enhancements Added in Version 1.100.4.0
- Modifications to not allow the user to add new VPWs or edit the outbound dial address
or line ID of any existing VPWs (CN2614)
Defects Resolved in Version 1.100.4.0
- A device with no appearances to register can show the SIP server icon as green instead
of black (F0013206)
- In the call register menu the clear all logs option should be greyed out when there
are no entries available in any of the logs (F0013217)
- The message waiting indicator does not operate correctly on an Asterisk PBX if the
PBX is configured to send out an extension number in the message account field (F0013240)
- Rapidly placing a call on and off hold repeatedly causes the unit to lock up (F0013244)
- When the currently selected page is set to read only, adding a key using key finder should
automatically change to the next page where a key can be added (F0013253)
- When outbound dialling entering a very long string (greater than 246 characters), with an
@ symbol early on in the string, crashes the UI (F0013260)
- For the dial tone locale some tones and frequencies are incorrect according to the ITU
spec (F0013271)
- When in the password entry box, moving the cursor one place from the end overwrites the
existing character when new characters are entered causing the password validation to fail
(F0013286)
- When a device is configured with DHCP disabled, the Network Status page shows "disabled"
where as other disabled settings show "Disabled" (F0013298)
Known Defects/Issues in Version 1.100.4.0
- With a large network latency and dialling a number that also rings on the calling iD808
the call state icon changes from connecting to established even though the call has not
been answered (F0013266)
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New Features/Enhancements Added in Version 1.100.3.0
- The operation of the clear speaker (assign-assign) has been enhanced to speed up the
user interface (CN2612)
Defects Resolved in Version 1.100.3.0
- Correction to internal I2C bus timings to the TI audio codec IC to match the latest
recommendations from TI (F0012751)
- When dialling an external number that is engaged, the iD808 initially plays a ringing
tone before changing to an engaged tone (F0012955)
- The DTMF tone is too loud (F0012985)
- G711 uLaw audio is transmitted at half amplitude (F0012985)
- The UI_CURRENT_USER_ID tag should be reported as blank to i cms in messages
sent when the iD808 is logged out (F0013034)
- The second ethernet port is configured as disabled by i cms but is enabled on
the iD808 device (F0013044)
- With an iD808 configured for two handsets and only one hidden privacy appearance,
if the user dials a number and selects privacy and the call is not answered and then
selects the other handset and repeats the same action the user should be blocked
from requesting privacy for the second call because the unit is configured for only
one private call (F0013049)
- The dialplan does not work for bridged call appearances (F0013067)
- When a SIP call is cleared immediately after applying or clearing privacy then the call
can sometimes not clear down properly (F0013087)
- Unpredictable behaviour can occur when taking a conference off hold and onto the handset.
One device shows handset busy message but the handset is empty and another device works
fine (F0013104)
- If a user has configured a dynamic key on an expansion module but is then seated on a
device without expansion modules then ringing calls are placed on the dynamic key on
the expansion module and so cannot be selected (F0013105,F0013142)
- The value of the i cms port number is incorrect in the M_PROFILE section of messages
from the device to i cms (F0013107)
- The names of the fixed keys shown in the free key wizard list are incorrect (F0013124)
- Calls cannot be answered when in the admin login screen (F0013125)
- The device cannot be logged in from i cms when in the safemode menu screen (F0013126)
- Speed dials which includes the & character in the name show the name as a blank string
against the key on the iD808 (F0013155)
- The option to add float keys is not available in the menu if there are no float keys
currently on the device and the current page being shown does not have any free paginating
keys (F0013177)
- When a directory entry is changed, the cache for the associated speed dial is not cleared
causing the display to still show the previous text until a resync or power cycle (F0013180)
- The key index is not set when creating a call by selecting a speed dial which can cause
problems when requesting privacy (F0013181)
- When in the key finder screens on an empty page, exiting out of key finder does not
change to a populated page (F0013187)
- Errors logged when far end ringing SIP call is move from a handset to a speaker channel
that previously had an established call that was later cleared (F0013207)
Known Defects/Issues in Version 1.100.3.0
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New Features/Enhancements Added in Version 1.100.2.0
- Kernel message logging to the messages file enabled to provide enhanced debug capability
Defects Resolved in Version 1.100.2.0
- When initiating an MRD call from an appearance with the handset configured for push-to-talk
the handset microphone is active despite the handset finger showing the handset as muted.
If the MRD is then placed in a conference etc, the whole conference has an active microphone.
If the handset button is pressed the error is corrected. Also an MRD call can be in the
idle state when on the handset when assigned from a dynamic speaker channel and the handset
is muted. Also the handset mute state is cleared when clearing a call on the handset when
configured for push to mute (F0012983)
- The i cms icon stays yellow after unseating a user from an iD808 that has
not got enough float keys (F0012989)
- When an Avaya SIP call that is a handset and is private is moved to a dynamic speaker
the icon on the associated appearance does not show the speaker channel icon and shows
the call as busy elsewhere instead of connected (F0012992)
- After creating a 3-way conference on a handset and moving it to a dynamic speaker channel
the dynamic speaker channel key still states 3-way even after another call has been added
to the conference (F0012993)
- When attempting a speed dial with a call already on the handset and Auto Hold off
you correctly get a message stating that the handset is busy. If you attempt a speed
dial when a conference is on a handset there is no error message (F0012996)
- Within the UI Alert Settings menu, Alert Override when set to Force Off should grey out
Ring On Busy. Currently it only greys out Ring Displayed Page (F0013005)
- Ring displayed Page options should be changed to On and Disabled, not On and Off as this
is confusing (F0013018)
- When the DHCP setting is changed from i cms but the same IP address is used then
the device gets into a state where changes made from the device are not sent to i
cms. This includes logging in and logging out. Updates can still be received from i
cms (F0013031)
- Sometimes the outgoing transmit audio is incorrectly muted after making a telephony
call private. The UI shows the call as unmuted and the fault can be cleared by muting
and unmuting the microphone (F0013032)
- When repeatedly pressing a blank soft key other key presses are delayed (F0013036)
Known Defects/Issues in Version 1.100.2.0
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New Features/Enhancements Added in Version 1.100.1.0
- Support for one or two iE816 expansion units
- UI page update speed improvements using graphics caching
- Float keys supported
- Paginating keys populated from left to right
- When rapidly paging only the page titles are updated until the user pauses
- * or # keys pressed immediately after a line seize now enters outbound dialling
- UI improvements to the transfer process
- Volume soft key introduced for handset/headset volume adjustment
- VAD mode for telephony calls is now configurable (Silence suppression is supported with
the G.729 and G.711 codecs but not with the G.722 codec)
- Latching mode for speaker channels is now configured against the speaker key instead of the
associated appearance key
- Country configuration for telephony tones
- Ethernet ports NET1 and NET2 are now configurable. When NET1 and NET2 are both enabled
the two ports can be used as a mini Ethernet switch for loop-through operation
- Japanese and English locales supported (but the offical release will be permanently
configured to use the English locale)
- Engineering screen capture utility added to filesystem
- Engineering packet capture utility added to filesystem
Defects Resolved in Version 1.100.1.0
- Miscellaeous fixes for the CDR output (F0012182, F0012188, F0012201, F0012682, F0012692,
F0012693, F0012706, F0012923, F0012924, F0012926)
- When using Avaya FNE's, a confirmation tone should be heard when the FNE is dialled (F0012338)
- Setup a call between an iD808 and a linksys phone with the iD808 preferred codec set to G722.
Press conference on the iD808 and then add another iD808 to the conference. Once the second
iD808 is added the voice drops in one direction between the linksys and the iD808s (i.e. no voice
from the iD808 to linksys) (F0012499)
- If a dynamic speaker key is configured with read only rights, when a call is assigned to it,
it can`t be cleared (by using assign-assign). The only way to clear the call is to wait for the call
to be cleared from the far end or resync or repower the unit (F0012520)
- When a hoot appearance is selected, the LED of the appearance is not updated. All other call
types change the LED to green (F0012530)
- When receiving an incoming call with the ringing shown on a dynamic finger if the menu is displayed
and the user answers the call by selecting the dynamic finger, the dynamic finger continues to
ring indefinitely (F0012531)
- When trying to edit a speed-dial that is linked to an address in a personal directory entry
which is not the default address, the address displayed in the edit dialog is the default
address instead of the selected sub-entry. Saving the key results in the default sub entry
being displayed on the key. When the unit is powered on and off or resynchronised it displays
the correct sub-entry again (F0012544)
- When the UI is showing a subpage from a page other than page 1 and that sub-page for page 1
is empty, re-synchronising from i cms doesn't show the page keys correctly (F0012554)
- When using the iD808 key programming menu to move a fixed key to a paginating key another
paginating key can be overwritten (F0012622)
- Call is not put on hold sometimes when a 3 way conference is put on hold and one of the
calls is cleared from the far end (F0012650)
- Can end up with one way voice when taking a call off-hold on a Cisco PBX (F0012660)
- When a device is logged out while playing a tone on the handset or speaker, the tone is not
cancelled and stays there while the device is logged out and when the device is logged in
again. The tone is only removed when a new tone is played on the same handset/speaker and
then cleared (F0012710)
- If a Caution dialog box is exited from manually and then the menu accessed, then the menu
is exited automatically after about a second (F0012716)
- The iD808 incorrectly allows the entry of the characters " < and > in label fields. These
characters are deemed unsafe for use in i cms and hence are blocked by i cms (F0012731)
- The tone played on the handsfree speaker (e.g. dial tone or far end ringing) continues to
be played out of the speaker when the call is moved to the other handset (F0012749)
- When fixed or paginating keys are added, deleted or modified, the local copy of the xml file
is not updated so re-powering the device returns it to the same state as the last synchronise.
Synchronising again updates the xml files and restores the keys to their state before the
power cycle (F0012756)
- When a device that is logged in is reconfigured from i cms the device shows that it is out-of-sync.
If that device is re-powered it is still out-of-sync but the i cms status icon on the
device is show as green (F0012768)
- When using a speed dial to dial the second leg of a transfer, the label of the seized line is
not updated. It still shows the name of the line and not the number or name dialled. The same
occurs when using the directories to dial (F0012769)
- When in the admin login screen the left and right nav keys do not move the cursor. An error
messages is logged and the cursor position is now confused making it difficult to delete or
complete entering the password (F0012800)
- Created a new speed dial key from the iD808 using the default appearance, but when going back
and editing this key it looks like it is using the appearance explicitly as opposed to using
the default appearance (F0012810)
- When editing a speed dial key that is linked to a personal directory entry, the iD808 always
send out the personal directory even when nothing has actually changed (F0012811)
- The iD808 is not restarted after changing the network configuration in safe mode. (F0012822)
- The Program Alerts menu shows "1-Alerts Settings". It should read "1-Alert Settings" (F0012826)
- After editing ARD, MRD and Hoot Speakers keys, the address label is cleared for any subsequent
call. This is seen when assigning speaker channel call to handset in two line mode (F0012828)
- The "£¥€" characters are missing as character options when adding or editing label fields from
the iD808 (F0012839)
- When adding a hoot appearance or speaker channel the address label is not set, resulting in
the second line being blank when putting the call on the handset (F0012879)
- Error messages seen in the log file when moving a connecting call to a dynamic speaker channel
and pressing the speaker channel key to enable the microphone (F0012890)
- An MRD or ARD call in the requesting privacy state can be added to a conference. This call
can then become private when no other local users are in the call resulting in the call
remaining private until the conference is cleared (F0012894)
- Soft keys are not updated when taking a conference off hold by pressing a speaker channel (F0012901)
- The first speaker channel can be incorrectly overwritten and assumed to be part of a conference
if an ARD, MRD or hoot call that is part of a conference has a state change apart from being ended (F0012902)
- When attempting to move a hoot speaker channel between handsets occassionally the handset busy warning box
appears when there is nothing on the destination handset (F0012925)
- Confirm actions have invalid title of "Project-Id-Version: iD808" for Call Register
options menu like clearing all log, clear log, copy entry and delete entry (F0012929)
- An update is sent to i cms when a speed dial or VPW is viewed even though no change has been made (F0012934)
- The clear speaker option in the main menu is sometimes incorrectly highlighted or greyed
out when calls are activated or cleared with the menu visible (F0012935)
- A ringing ARD/MRD call will not be shown on a dynamic key if the dynamic key already contains
another call and this call is cleared (or answered) (F0012946)
Known Defects/Issues in Version 1.100.1.0
- When initiating an MRD call from an appearance with the handset configured for push-to-talk
the handset microphone is active despite the handset finger showing the handset as muted.
If the MRD is then placed in a conference etc, the whole conference has an active microphone.
If the handset button is pressed the error is corrected. Also an MRD call can be in the
idle state when on the handset when assigned from a dynamic speaker channel and the handset
is muted. Also the handset mute state is cleared when clearing a call on the handset when
configured for push to mute (F0012983)
- Sometimes the outgoing transmit audio is incorrectly muted after making a telephony
call private. The UI shows the call as unmuted and the fault can be cleared by muting
and unmuting the microphone (F0013032)
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New Features/Enhancements Added in Version 1.010.8.0
- No features or enhancements added to this release
Defects Resolved in Version 1.010.8.0
- Sometimes the outgoing transmit audio is incorrectly muted after making a telephony
call private. The UI shows the call as unmuted and the fault can be cleared by muting
and unmuting the microphone (F0013032)
Known Defects/Issues in Version 1.010.8.0
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New Features/Enhancements Added in Version 1.010.7.0
- No features or enhancements added to this release
Defects Resolved in Version 1.010.7.0
- When initiating an MRD call from an appearance with the handset configured for push-to-talk
the handset microphone is active despite the handset finger showing the handset as muted.
If the MRD is then placed in a conference etc, the whole conference has an active microphone.
If the handset button is pressed the error is corrected. Also an MRD call can be in the
idle state when on the handset when assigned from a dynamic speaker channel and the handset
is muted. Also the handset mute state is cleared when clearing a call on the handset when
configured for push to mute (F0012983)
Known Defects/Issues in Version 1.010.7.0
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New Features/Enhancements Added in Version 1.010.6.0
- No features or enhancements added to this release
Defects Resolved in Version 1.010.6.0
- SIP calls cannot be made private if an ARD, MRD or hoot call is added to a conference.
The fault is only rectified by re-synchronising or re-powering the unit (F0012959)
Known Defects/Issues in Version 1.010.6.0
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New Features/Enhancements Added in Version 1.010.5.0
- No features or enhancements added to this release
Defects Resolved in Version 1.010.5.0
- Sometimes audio is not heard when accessing voice mail on an Avaya PBX (F0012938)
Known Defects/Issues in Version 1.010.5.0
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New Features/Enhancements Added in Version 1.010.4.0
- No features or enhancements added to this release
Defects Resolved in Version 1.010.4.0
- The microphone is active when it should be muted when a SIP call is moved to the
dynamic speaker channel before it is answered. Also when an incoming SIP call is
placed on a dynamic speaker channel (and the microphone is muted) and subsequently
a user barges in to this call the call on the dynamic speaker is no longer muted
however the dynamic speaker still indicates a muted microphone (F0012884)
Known Defects/Issues in Version 1.010.4.0
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New Features/Enhancements Added in Version 1.010.3.0
- No features or enhancements added to this release
Defects Resolved in Version 1.010.3.0
- When synchronising a device from i manager sometimes the device logs out
and remains logged out. Any configuration update from i manager would not
have happened and the device will need to be synchronised a second time (F0012712)
- Very occassionally an MRD channel may not ring and will remain in this state
until the device is re-powered (F0012798)
- If the total of hoot, MRD and ARD channels attached to keys is more than 30, some of
the channels may not be initialised correctly and hence will not work (F0012867)
Known Defects/Issues in Version 1.010.3.0
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New Features/Enhancements Added in Version 1.010.1.0
- Login and resynchronise times speeded up. As a result of this change the version
of imanager should be 1.110.x.x or later (CN2599)
Defects Resolved in Version 1.010.1.0
- Silence is sent to the voice recorder when one user drops out of a conference (F0012646)
- The iD808 does not list calls on hold, either via the hold or conference buttons in the CDR
heartbeat event. This resuts in any on hold calls being considered as ended by the voice
recorder when the CDR heartbeat event is sent (F0012651)
- A paginating MRD call appearance can incorrectly generate the ringback tone on the handset
when it is selected. This only occurs when the internal index used to reference the
appearance matches the index of an ARD speaker channel (F0012686)
Known Defects/Issues in Version 1.010.1.0
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New Features/Enhancements Added in Version 1.0.15.0
- The screen saver time period has been increased from 1 hour to 2 hours (CN2580)
- Assign-assign followed by a static ARD speaker channel key can now be used to clear down a
call on a static ARD channel as an alternative to assigning the call to a handset and
clearing the call from the handset (CN2587)
- MRD calls on speaker channel now do not transmit SbRTP when the gooseneck microphone
is muted (CN2587)
- MRD calls on speaker channel now show alerting when incoming ring is received unless
they are active talkers (CN2587)
- The privacy operation for ARD and MRD has been modified so that a user that selects
privacy only makes the call private for the local users and does not affect the privacy
state of the far end users (CN2590)
Defects Resolved in Version 1.0.15.0
- When making attended transfers on an iD808 and clearing down the second leg to go back to
the first leg sometimes the first leg is cleared (F0011401)
- When the line seize times out while transferring a SIP Call on an Avaya PBX sometimes you cannot
cancel out of the transfer dialog. Also in this situation if no appearances are available the
user should be warned that the call is being returned to the handset (F0011792)
- When an ARD call on a handset clears from the far end there is a period of time where the
handset is shown as busy elsewhere and during this period cannot be used to answer another
call (F0011854)
- If a telephony call on a speaker channel is assigned to a handset that has privacy pre-selected
the privacy padlock remains grey and privacy cannot be selected or cancelled for that call (F0012049)
- If you log a user off an iD808, remove the network cable and log the user back in the i
cms icon is red as expected. When the network connection is replaced the i cms icon remains
red (F0012052)
- When using an ARD there is approximately a ten second delay between answering and it becoming
active if VAD is enabled (F0012074)
- With Simplex Local Muting enabled there is a burst of audio from a channel when it`s enabled
and disabled whilst another call is active (F0012078)
- The iD808 incorrectly uses the caller ID from an INVITE message instead of a NOTIFY message
when the call is presented on a dynamic key and this can cause an inconsistency for the caller ID
label between a call shown on an appearance key and a call shown on a dynamic key (F0012082)
- Occasionally a process is seen to quit with an internal error whilst accessing configuration
data. Only reported on one iG330 unit but the bug is in common code that is used on the iD808
so in theory the problem could occur on an iD808 (F0012092)
- Make a telephony call and move the call to a dynamic speaker channel. Clear the call from the
far end and some error messages are logged. This does not seem to cause any other problems (F0012100)
- Turning DHCP off may fail. This bug has been noticed in the iG330 and since the error was in
code that is common between the iG330 and the iD808 it is possible that the problem could also occur
in the iD808 although this has not been noticed yet (F0012101)
- When an appearance key is in the call failed state the key will not indicate a new call received
until the failed state timer finishes (F0012102)
- When a voice recording stream is configured for an IP address on the local subnet that is not
contactable other DSP actions such as start and stop ringing can be delayed by several
seconds (F0012106 & F0012418)
- If the Push to Talk/Push to Mute button on the handset is pressed whilst the "Handset is Busy"
error message is displayed then the handset mode becomes latched in the push position (F0012121)
- The values entered for the SNMP in the network configuration page are not saved after the unit
logs off and back on. i cms overwrite all the parameters with the ones set in iManager (F0012134)
- A call appearance is shown as being permanently in use after dialing another user, pressing privacy,
then clearing the call before the far end answers (F0012135)
- Inbound and outbound dial string conversion does not operate correctly (F0012136)
- Enabling a hoot speaker channel clears the privacy requesting and privacy status of the selected
handset. The handset icon does not get updated so the call still looks private until it is move
between handsets or on to a speaker channel. At this point the call is still private but the
handset icon is not private (F0012156)
- The dial plan should only be applied to those appearances that are attached to the same PBX
as the default appearance (F0012168)
- If a listen only hoot is assigned from a speaker channel to a handset and then cleared from the
handset the speaker channel icon still shows the handset as attached (F0012169)
- Miscellaeous fixes for the CDR output (F0012182, F0012189, F0012190, F0012198, F0012211, F0012467)
- Softkeys are not updated when moving a hoot speaker channel between handsets (F0012209)
- A handset does not remain muted for a listen only hoot channel when the handset button is pressed
then released (for PTM) or the call is moved to the other handset. No audio is sent out on this
channel but the audio is sent to the voice recorder (F0012220)
- Unable to seize a line, call appearance or bridged call appearance with a line appearance set as
the default if there are call appearances owned by the user that are not attached to any keys and
have a long label that is first in the list of all call appearances when sorted alphabetically (F0012223)
- An incoming call that that is cleared down before the iD808 UI shows the call as ringing can
remain ringing until the user tries to answer the call, at which point the call will be shown as
a failed call (F0012249)
- When dialling a call from the handset and the using the handsfree option it is not possible
to change the ringback volume level with the master volume control (F0012250)
- Using Avaya, when an incoming call rings on a dynamic key and is answered by pressing the dynamic
key the call is correctly answered but the dynamic key continues to show a ringing call (F0012251)
- When making a call and clearing it down again very quickly there can be entries in the redial
menu that are not correctly formatted (F0012253)
- Softkeys are not updated when the second leg of a transfer is cleared from the far end (F0012261)
- A login attempt from an iD808 may time out if the profile being downloaded from i cms is
large (F0012265)
- With an MRD on a speaker channel it is not possible to distinguish between a call that is active
but has the gooseneck microphone muted from one that is inactive (F0012270)
- An error message is logged when enabling a hoot appearance on a dynamic speaker channel (F0012271)
- Sometimes a click is heard on the iD808 when pressing the clear button to clear a hoot
channel from a handset (F0012283)
- When no more keys can be added because the 600 paginating keys limit has been reach the
key programming user interface still allows the user to try and add another paginating key (F0012292)
- MRD and ARD calls do not ring correctly if they were previously cleared when requesting privacy.
The ring icon is not a moving bell but a static bell with a grey padlock (F0012309)
- An MRD call ringing on a dynamic speaker channel always shows the style for profile 1.
Also when an MRD call on a dynamic speaker channel is in the busy elsewhere private state,
the channel can still be selected and goes to the connecting state. Also enabling an MRD on
a dynamic speaker channel takes the privacy requesting state from the selected handset (this
only affected interim code and was not a problem with any official release) (F0012310)
- After moving an ARD or MRD appearance key from a non-paginating key to a paginating key or vice
versa their labels are displayed incorrectly when their call is moved to a handset or dynamic
speaker channel (F0012321)
- When using the Calling Number Block FNE followed by dialling an external number the UI
crashes (F0012333)
- When dialling an external telphone number that is engaged, no engaged tone is heard on the
iD808 (F0012359)
- The iD808 can lock up when receiving SbRTP traffic from i-series devices operating in
v2.3 mode (F0012366)
- No ringback tone is heard when assigning a connecting ARD to the handset and then pressing
handsfree. If the handsfree button is toggled the ringback tone is then heard. Also with
handsfree selected, assigning a connecting ARD to handset plays the ringback tone but the
volume cannot be adjusted (F0012372)
- If RTCP is enabled it is possible that RTCP may stop transmitting and it is possible that
ARD calls will not clear down correctly (F0012373)
- If two ARD callers are in the connecting far end ringing state, they can hear each other.
If one user puts the call on handsfree audio the ring back tone is correctly heard on the
main speaker. If the handsfree button is selected again then the ring back tone returns
to the handset but the audio received still comes out of the main speaker (F0012376)
- During a 3-way conference move the conference to the other handset (select the other
handset and then press one of the active conference appearances). Then move the conference
back to the original handset but use the other appearance to do so. Although the conference
appears to have moved back to the original handset all the audio streams continue to go
through the second handset (F0012378)
- Unattended transfer can cause the first leg of the call to drop if you press transfer
very quickly after dialling the second leg (before the 2nd leg far end rings) (F0012385)
- When clearing an ARD speaker channel that is on-hold using assign+assign the text is removed.
This also occurs if the ARD is part of a conference and the conference is on hold
(this only affected interim code and was not a problem with any official release) (F0012400)
- A strange tone is heard from main speaker when a SIP call is cleared from the far end and at
the same time a new incoming call starts (F0012401)
- UI crash because of access violation. This was only observed in development code that had a
different memory map to released code (F0012407)
- If an iD808 is configured with an Avaya bridged call or line appearance and no call appearance
(an invalid configuration) the appearance key flashes while trying to seize a line but the
user is not informed that the unit is not configured correctly (F0012422)
- After assigning a hoot channel to a conference and clearing it down audio is still heard on
a hoot speaker channel but the VAD indicator is always off (F0012433)
- Hoot speaker channel do not work correctly (this only affected interim code and was not a
problem with any official release) (F0012436)
- Cannot clear an MRD call from a handset (this only affected interim code and was not a
problem with any official release) (F0012443)
- Non-latched Hoot, ARD and MRD calls on dynamic speaker channels cannot talk on the gooseneck
microphone (this only affected interim code and was not a problem with any official release) (F0012459)
- Global muting does not work for MRDs on dynamic speaker channels (this only affected
interim code and was not a problem with any official release) (F0012461)
- A unit configued without an Avaya appearance reports the error "No valid master appearance"
(this only affected interim code and was not a problem with any official release) (F0012462)
- ARDs cannot connect again after going into the active microphone muted state on a fixed
speaker key (this only affected interim code and was not a problem with any official release) (F0012468)
- ARD calls on dynamic speaker channels in the busy elsewhere private state can still be
selected and change to the connecting state (this only affected interim code and was
not a problem with any official release) (F0012470)
- The UI can lockup when an upgrade is initiated from i cms immediately after the iD808 is
powered up (F0012471)
- When upgrading an iD808 the old iD808_upgrade_*_tar.gz files in the upgrade directory are
not deleted (F0012472)
- When an ARD call is in the active microphone muted state and another user presses privacy, a
few packets are sent out before the call drops to the busy elsewhere private state causing
the other user to stop transmitting privacy (this only affected interim code and was not
a problem with any official release) (F0012473)
- In the Key Definition Help text, the page key is listed with a capital P however the page
key has a lower case p (F0012474)
- Auto handset mute does not work when moving an active ARD appearance from a dynamic speaker
channel to the other handset by pressing the appearance key or when talking an ARD call
off hold (F0012485)
- When an ARD call is in the requesting privacy state (because other local desk stations are
active) and assigned to a speaker channel then the requesting privacy symbol stays on the
channel however the channel is no longer attempting to become private (this only affected
interim code and was not a problem with any official release) (F0012490)
Known Defects/Issues in Version 1.0.15.0
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New Features/Enhancements Added in Version 1.0.14.0
- CDR change to include IP address in message header and changed CDR version to 4.0 (CN2579)
Defects Resolved in Version 1.0.14.0
- UI crash in the following sequence of events. Make a call using a call appearance that
currently resides on a key adjacent to a speaker channel volume control. Answer the call
at the far end. Make the call private. Move the call to a dynamic speaker key that is below
the call appearance key. Copy the call back to the handset. Attempt to make the call private
and the UI crashes (F0012072)
- When there are multiple talkers on an ARD channel and the call is cleared down from
the far end the iD808s continues to transmit the SbRTP stream from each of the
multiple talkers until those iD808s re-connect to the ARD channel and clear down
normally. This can result in the talker limit being reached as a result of some iD808s
incorrectly transmitting when they should be idle (F0012085)
- The CDR event is incorrect when the ARD channel on multiple deskstations are active and the
ARD call is cleared from the far end. This causes the iD808 to generate the
ANSWERED_BY_LOCAL_END event and not the CALL_ENDED event (F0012096)
- The CDR heartbeat event checks for active calls by looking at the should record flag
for the call. This should only be set when a CDR call event is sent however some calls are
hidden from the CDR (SIP features calls) but these calls still get the should record flag
updated and can be listed in the heartbeat events (F0012098)
- Missing CDR events when an ARD call in a conference is cleared from the far end. The
CDR_CONF_STATUS event is missing when the ARD call is cleared and the CDR_CALL_ENDED event
is missing if the ARD call is the last call in the conference (F0012114)
- When clearing the second leg of a transfer before the second leg is established using the
clear key, the CDR Link Event shows Audio = 1. This should be Audio = 0 to indicate
no audio for the second leg (F0012122)
Known Defects/Issues in Version 1.0.14.0
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New Features/Enhancements Added in Version 1.0.13.0
- Local and global muting functionality has been added. Previously there was some local
and global muting support but this has been significantly revised for this release to
meet the market requirements (CN2566)
- The call state for an MRD speaker channel is now always shown as connected (CN2566)
- Change in ARD behaviour so that ARD calls on speaker channels that are on-hold or busy
elsewhere (including ringing) are not heard on the speaker and do not illuminate the
VAD indicator (CN2566)
- CDR protocol modified to match revised CDR specification (CN2564)
- Handling of inbound calls enhanced to allow the caller ID to be adapted to handle different
dial string formats (CN2575)
- Handling of outbound calls enhanced to allow the dial string of speed dial, call register and
directory entries to be adapted to handle different dial string formats (CN2575)
- Core files are now generated if any application crashes
Defects Resolved in Version 1.0.13.0
- Very occasionally an Avaya subscription fails after a resynchronisation from i cms
and the SIP status icon is yellow (F0010680)
- With the handset muted, place a call on a dynamic speaker and then assigned it back to the handset.
Press assign twice followed by the speaker channel key to clear the call off the speaker. The microphone
indicator remains active (F0011361)
- If an iD808 is configured to use a static IP address and this address is changed from i cms,
i cms loses communication with the device until the device is repowered or a configuration
change is made on the device (F0011379)
- When making attended transfers on an iD808 and clearing down the second leg to go back to the
first leg sometimes the first leg is cleared (F0011401)
- If a subcription tries to refresh when the subscription server is not accessible on the
network the subscription fails and does not recover when the server is available again. The
SIP status icon is yellow (F0011492)
- In the CDR event generated when putting a call on-hold or taking off-hold the values for audio
microphone and audio speaker are incorrect. (F0011615)
- Asterisk call un-hold and Avaya privacy call (going private) sometimes has delayed RTP media
re-connection (delay > 5 secs) on the handset. (F0011698)
- The ARD or MRD ringing LED is cleared if the call is on a speaker channel and audio is detected.
The LED changes to report the voice activity and not the ringing. (F0011712)
- With a conference on a dynamic speaker channel containing two SIP calls and an ARD call on a
speaker channel, when audio is present on the ARD call the voice activity LED comes on for
this call but not the conference voice activity LED. (F0011713)
- When a 3-way conference is moved to a dynamic speaker channel and then moved back to a handset
press the conference button to put the conference call on hold and then press the dynamic speaker
channel button to unhold the conference on the handset. Now, if you clear the dynamic speaker
the conference is disconnected where it should have only detached the conference from the dynamic
speaker. (F0011716)
- Depending on the order of talk activity within an SbRTP conference that the iD808 is connected
to with an active link, there can be some leakage of audio buffers. After some time,
this could lead to loss of receive audio from new talkers within any SbRTP conference
(hoot or ringdown), or loss of receive audio for any SIP call within this unit (F0011742)
- Connect an ARD call on a handset. Press the conference button twice to put the ARD call into
a conference call. Now clear the ARD call from the far end. The call is disconnected from the
far end but is still shown as connected on the handset finger. (F0011747)
- Select privacy on the selected handset and make an ARD call using an ARD appearance. The
ARD call appears to immediately connect even though the far end has not rung or answered
and the padlock remains in the requesting state (grey padlock shown) (F0011748)
- RTP Payload Code changes in i cms are not acted upon by the iD808 when the device is
synchronised (F0011751)
- On an Avaya PBX dialling an extension that is set to call forwarding ends up connecting to the
call forwarded number but the UI still shows the original number that was dialled (F0011752)
- User A dials user B and user B does not answer. When you look at the missed calls list, 2
entries are present for 1 missed call (F0011778)
- Go to the personal directory, press the left arrow to see view details. Press ok again to
see details. Press ok again to see details. Press the left arrow to go back and the text
has reverted to small text. Should be big text (F0011783)
- If a line seize times out while transferring a SIP Call on an Avaya PBX you cannot cancel out
of the transfer dialog (F0011792)
- Prompt bar text for the "OK" key changed from "Select" to "Select/Chg" for the directory edit,
key edit, config edit, login and ping menus (F0011801)
- The iD808 is configured with an invalid netmask when changing to a static IP address
configuration (F0011819)
- If an Avaya PBX line seize fails and you try seizing another line very quickly which doesn't
fail the new line is shown as busy elsewhere but actually it is not. To clear the problem
seize the line and then clear it (F0011820)
- ARD/MRD speaker channel audio is incorrectly transferred to the handset when there is an attempt
to assign the channel to the handset when in the busy elsewhere private state (F0011830)
- After a speed dial is added to a key from a corporate or personal directory and before the
unit is repowered or resynchronised the speed dial key can show alerting for an incoming
call (F0011843)
- An iD808 that has no default appearance set but does have appearances set on the device should
show the i cms icon as yellow but it stays green (F0011845)
- When an ARD call on a handset clears from the far end there is a period of time where the
handset is shown as busy elsewhere and during this period cannot be used to answer another
call (F0011854)
- If the privacy key is pressed when the handset is idle and if there is no hidden handset
appearances configured for the iD808 the user is blocked from requesting privacy in advance
of making a call with the message "Privacy call is not enabled" (F0011855)
- With an iD808 configured with no hidden handset appearances and receiving an incoming call to
an Avaya call appearance the finger rings but the caller ID is blank (F0011858)
- DTMF tones not being recognised when using uLaw. Volume level of tones needs to be higher (F0011878)
- When the iD808 receives notification of an incoming call via a NOTIFY message with a blank
display name the caller ID is shown as blank on the ringing appearance key. This can happen
with external calls on an Avaya PBX (F0011895)
- Speaker key icons are not correctly updated when making or clearing privacy from calls (F0011912)
- SIP Users appearance fails Registration to PBX when PBX acknowledges the response "423
Interval Too Brief In Expires Header" with its minimum expiry value. Any subsequent
registration request will also fail. Re-syncing may fix the problem (F0011916)
- Two different ring-back tone are heard during a SIP call transfer using an Avaya PBX (with
Mu-law Codec) after the previous SIP call transfer was cancelled by the transferer. The
transferer had cancelled the SIP call after it was answered by the far end. The second
ring-back tone was heard in the media stream sent by Avaya PBX (F0011923)
- Incorrect voice recording profile during startup. Some of the voice recording stream connections
are broken (F0011930)
- The caller ID should not be checked when attempting to match incoming calls to VPW lines (F0011934)
- A SIP call on the dynamic speaker is incorrectly cleared by pressing the clear button of the
selected handset in the following scenario: - Make a SIP call and move it to a dynamic speaker
channel, move the call back to the selected handset and press privacy to enable it. Press
privacy again to disable privacy. Clearing the call will clear the call from the handset
instead of moving it back to the dynamic speaker channel (F0011937)
- Editing a VPW key speed dial entry using the personal directory corrupts the VPW labels (F0011938)
- When in the menu screen conferences cannot be moved between handsets. Calls can be moved
however if they are on a dynamic speaker channel but the icon is not updated (F0011948)
- If in the future a new element or attribute is added to one of the XML files used on the iD808 and
then i cms is upgraded without upgrading all the iD808 being controlled from that i cms
the iD808 will reject the XML file because its DTD for the file does not match. This will leave
the iD808 in a non-working state until it is upgraded. This is unacceptable because the entire
upgrade process may take several days (F0011953)
- When the device attempts to open the i cms listening socket after it has just been closed then
it fails. This can lead to changes made from iManager not being applied because the connection
has been refused (F0011969)
- When moving a private SIP Avaya call to the speaker channel the gooseneck microphone is not muted
whereas it is shown muted (F0011972)
- When using handsfree and push to mute on the handset, if you mute the handset during handsfree
mode you then cannot press the handset key button to mute hansdfree (F0011981)
- With a telephony call on a dynamic speaker channel and then assigned to a handset and the second
handset selected and in the idle state pressing assign followed by the dynamic speaker key
correctly ignores the key press but an "Action not possible" message is not displayed (F0011985)
- Global muting does not work for ARD and MRD appearances (F0011988)
- If a speaker channel is on a handset and the other handset is selected, when the user presses
assign followed by the speaker channel then the message Action not possible appears and then
the channel changes to the other handset. The Action not possible message appears on the button
press and the moving between handsets happens on the button release. The moving between handsets
should only occur with button presses and hence the button release event should have been
ignored (F0011992)
- When in the privacy pending mode, changes to global muting are not monitored so the channel can
remain muted or unmuted until the user activates the channel (F0011995)
- The alert profile does not take effect on the subsequent ARD/MRD incoming calls when the previous
ARD/MRD call was not answered and alert profile mode is set to ringing to be played after 1 to
14 seconds (F0011996)
- When the network connection is disconnected and a user on the iD808 logs in using the correct password
and then immediately logs out there are errors reported in the error log (F0012003)
- When an ARD call is on hold and picked up by another user, the wrong CDR event is generated and the
"should record" flag is set. No CDR event is generated when an ARD or MRD call on a dynamic speaker
channel is moved to the handset by pressing the appearance key. The wrong CDR event is generated
when a SIP call is answered on a bridged appearance (F0012007)
- If an ARD appearance is in the call failed state, it can still be moved between handsets in this
state. This causes the CDR data to be incorrect as the call ended event is generated followed by
recording stream info events with the "should record" flag set when there is nothing to
record (F0012010)
- If the MTFIF application starts before the CURRENT_DNS_SERVER configuration variable is set any
registration or subscription that requires DNS lookup fails. This has been noticed once during
an upgrade (F0012017)
- The iD808 does not send audio received to the voice recorder until we join the call for the first
time. This means, after starting up, we can be receiving voice on hoot and MRD channels and playing
it out of the iD808 speaker but not forwarding it on to the voice recorder. Joining the call corrects
the problem and exiting from the call still sends received audio to be recorded. This problem also
occurs when a hoot or MRD channel is added to a speaker key (F0012022)
Known Defects/Issues in Version 1.0.13.0
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New Features/Enhancements Added in Version 1.0.12.0
- No features or enhancements added to this release
Defects Resolved in Version 1.0.12.0
- The DTMF tone for digit 9 may not be recognised by the receiving party (F0011378)
- iD808 to iD808 calls attempting to use G722 with the Avaya PBX end up with calls using
G711 (F0011564)
- With the iD808 hosting a conference set to Push-to-Talk handsets and with 3 iD808’s in a
3-way conference the first phone dialled and added to a conference by the initiating iD808
loses voice to it when the second phone is added to the the conference (F0011670)
- If a voice recording stream is enabled but the IP address is left at 0.0.0.0 the unit
sends traffic out to 0.0.0.0 (F0011674)
- Softkeys are not updated when moving an MRD speaker channel call between handsets (F0011682)
- When handsfree is activated in handset muted mode handsfree is correctly show un-muted but
the LED indicator status is wrong (F0011685)
- When going to the admin login screen and then exiting again without entering in the password
some strange things can happen where the device does not think a user is logged in. An example
of this is when viewing the online help pages, incoming calls are ignored (F0011688)
- When configured for push to talk if you select handsfree and then go back to the handset
the handset is not muted even though the switch is not pressed (F0011689)
- When an ARD call is connecting the ringback tone is played on the handset. If handsfree is
selected and then the call is answered the ringback doesn't stop and continues to be played
on the speaker (F0011690)
- When an ARD is connecting and put on handsfree and taken off, the ringback tone is played
down the line (F0011692)
- When an ARD call is in the connecting state and the ringback tone is heard on the handset,
and this call is put on the handsfree speaker, when the call is answered the audio is on
the handset and not the handsfree (F0011695)
- When taking a call off handsfree the handset muting does not take into account the current
handset switch position (F0011696)
- Putting an ARD call on hold does not mute the microphone. The microphone LED is not active
but the microphone is active (F0011699)
- If a connecting ARD appearance is moved between handsets, the ringback tone is not moved.
If the call is answered on the other handset then the tone is not stopped (F0011700)
- Voice recording audio output for ARD's, MRD's & hoot's that are assigned to speaker channels
do not work until they have been assigned to a handset then removed. (F0011708)
Known Defects/Issues in Version 1.0.12.0
- Very occasionally an Avaya subscription fails after a resynchronisation from i cms
and the SIP status icon is yellow (F0010680)
- If an iD808 is configured to use a static IP address and this address is changed from i cms,
i cms loses communication with the device until the device is repowered or a configuration
change is made on the device (F0011379)
- If a subcription tries to refresh when the subscription server is not accessible on the
network the subscription fails and does not recover when the server is available again. The
SIP status icon is yellow (F0011492)
- In the CDR event generated when putting a call on-hold or taking off-hold the values for audio
microphone and audio speaker are incorrect. (F0011615)
- Asterisk call un-hold and Avaya privacy call (going private) sometimes has delayed RTP media
re-connection (delay > 5secs) on the handset. (F0011698)
- The ARD or MRD ringing LED is cleared if the call is on a speaker channel and audio is detected.
The LED changes to report the voice activity and not the ringing. (F0011712)
- With a conference on a dynamic speaker channel containing two SIP calls and an ARD call on a
speaker channel, when audio is present on the ARD call the voice activity LED comes on for
this call but not the conference voice activity LED. (F0011713)
- When a 3-way conference is moved to a dynamic speaker channel and then moved back to handset
press conference button to put the conference call on hold. Then press the dynamic speaker
channel button to unhold the conference on the handset. Now, if you clear the dynamic speaker
the conference is disconnected where it should only remove the conference from the dynamic
speaker. (F0011716)
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New Features/Enhancements Added in Version 1.0.11.0
- When automatically seizing the first available default appearance key during manual dialling,
speed dialling, etc., the key chosen is the one with the lowest appearance number. This differs
from the previous behaviour which choose keys based on whether they were fixed or paginating
and their relative position in the key layout
Defects Resolved in Version 1.0.11.0
- When carrying out an unattended transfer the "transferred to" user see's the name of the person
transferring the call until the call is picked up then the name switches to the original initiator.
The name shown on the "transfer to" user should switch to the original originator before the call
is answered (F0010933)
- The user settings help text is not complete. The new user preference options available are not
listed (F0011024)
- After a unit has been powered on for approximately 74 hours ARD calls can be left in the busy
elsewhere state after the call is cleared down (F0011138)
- After using DebugDsp over a SSH session the iD808 UI can lock up (F0011223)
- The microphone active LED is not cleared if an active hoot speaker key is deleted from the device
using the menu (F0011364)
- With auto hold enabled and a conference on a handset containing fixed speaker channels, assigning
the conference to one of the speaker channels shows the "Action not possible" message but also
incorrectly puts the conference on hold (F0011370)
- The user is not informed if we receive a message from i cms but ignore it because we are
logged in. This can cause confusion as i cms will show the device as out of sync but the
device will still show a green i cms icon instead of yellow (F0011371)
- Synchronising a logged out device causes the i cms icon to go yellow. Entering safe mode
and looking at the Network Status page shows that the problem is because no default appearance
is available for this user (F0011374)
- With RTCP enabled for the recording streams, re-synchronising causes some speaker channels to have
a cross symbol. The iD808 had 11 speaker channels on it and the last one has a cross on it (F0011375)
- Place an active SIP call on a dynamic speaker and then delete the dynamic speaker key from the
iD808. The call is not dropped and has two way audio between the internal mic on the iD808 that
had the dynamic speaker channel and the handset on the other unit (F0011377)
- iD808 doesn't send DTMF tones to line when on handsfree and the microphone is muted.(Part of F0011378)
- When a key is being deleted and that key is part of the active conference call the conference
call becomes a dangling call and is not removed from the conference (F0011391)
- ARD speaker calls start off with the speaker muted. When in the busy elsewhere state the voice
activity indicator works but no voice is heard. Entering, then leaving the call corrects this
and the speaker is no longer muted (F0011394)
- Occassional one-way voice for some call legs in a conference (F0011403 & F0011620)
- When a second user was added into a conference all voice was lost between all 3 units and just
noise was heard (F0011405)
- When auto hold enabled on the iD808 and an active call on the selected handset, seizing a line
puts the first call on hold but the handset incorrectly shows the remote caller ID from the first
call on the handset key (F0011408)
- When dialling an anonymous call appearance and then cancelling the call before the call is answered,
the finger showing the call appearance status shows busy elsewhere and then eventually is closed
after a time out (F0011433)
- During power-up occasionally the DSP is restarted a second time which leaves the internal ethernet
switch in half duplex mode and the network status icon in the yellow state (F0011442)
- When making a call private and placing it on hold, close the call down from the opposite end.
Repeat these steps for as many times you have have hidden handset appearances plus one. For
example if you have two hidden handset appearances follow these steps three times. The last time
you try to press the privacy button you will receive an error message stating privacy is not
available (F0011447)
- Set up a call between 2 iD808's using G722. On one of these iD808s make another call
to another device using the second handset and using G729. On answering the second call
audio may be dropped one way on the initiating iD808 (F0011456)
- When using a static IP address the DHCP status is shown as "Unknown" in the Show Network status
screen (F0011461)
- When a MRD channel that is receiving audio is activated the light on that channel's button will
go out despite the channel still receiving audio (F0011470)
- The user should not be allowed to transfer a call that was delivered on an anonymous call appearance (F0011485)
- Add two telephony calls to a conference call. Put the conference on-hold and clear down the first call
added to the conference from the far end. Now when you try to make the conference call active the
conference call is dropped. (F0011495)
- When using asterisk and there is only a single call in a conference call no music-on-hold is heard
on the other end (F0011496)
- There is a potential race condition on start-up that could result in the internal MAC address variable
used for i cms communication not being initialised (F0011502)
- In the key definition help screen the key "i-com" is listed. This has now been relabeled "i" on the
latest mechanics and hence the help text should also be changed to "i" (F0011506)
- Using 2 iD808s with eight ARD channels on speaker keys, when making G729 calls between these units
and putting these calls on-hold and moving calls to dynamic speaker channels sometimes the calls
can end up with no audio (F0011535)
- With a round trip delay of greater than 300ms between the iD808 and the Avaya PBX making SIP calls
is unreliable and with a round trip delay of greater than 450ms SIP calls cannot be made (F0011544)
- Receiving a firmware download while in the admin password entry screen leaves the selected handset
indicator on (and any others). This is because the iD808 is not logged out before starting the
download (F0011551)
- Error message shown in log file when in the safemode menus and the device is syncronised from
i cms (F0011552)
- When an iD808 receives a SIP INVITE that only specifies G729 and includes "annexb=no", when the
call is answered the iD808 sends back silencesupp on which is illegal for G729 if annex B is not
supported. This can cause a incoming call to fail on an Asterisk PBX (F0011566)
- In a 3-way conference it is possible to assign both call appearances used to make the
conference to multiple handsets and multiple dynamic speaker channels. The unit also got into
a mode where the appearances of the conference could not be assigned to a speaker or
handset finger, although the conference was still functioning on the handset. The only way to
clear this situation was to close the calls at the other iD808's, therefore closing the
conference (F0011578)
- Set up a 3-way conference and place that conference on handsfree. The conference is still on the
handset. Press the handsfree button again and the conference is now on handsfree. i.e. - handsfree
operation is back to front (handsfree speaker LED indication ON when on the handset and handsfree
speaker LED indication Off when on handsfree) (F0011588)
- If multiple MRD calls are in a conference then pressing the ring soft key only rings one or
none of them (F0011603)
- The production test dsp files (sbdsp0prodtest.ldr & sbdsp1prodtest.ldr) are swapped with each other (F0011614)
- Add a speed dial to a key and then attempt to add a dynamic speaker to another key. When save is
pressed an error message is logged and the key press is ignored so that the dynamic key cannot
be added (F0011623)
- If a call is on a handset and the handset is muted and then the handsfree button is pressed the
handsfree microphone is not muted. If the handset is unmuted then muted again then the handsfree
microphone is now muted (F0011627)
- With no call on the selected handset the transfer button was pressed and the error message "Cannot
transfer barged in call" was shown. This occurred on both handsets and could not be cleared by
pressing clear or adding and clearing calls from handsets. Could not reproduce a second time (F0011628)
- When changing the alert settings of a paginating key using "Alert key" or "Alert page" the changes
do not take affect until the page is redrawn (e.g. by moving away and back to the page again using
the nav keys) (F0011634)
- Miscellaneous fixes to the CDR output (F0011637, F0011639, F0011642, F0011648, F0011652 & F0011653)
Known Defects/Issues in Version 1.0.11.0
- Very occasionally an Avaya subscription fails after a resynchronisation from i cms
and the SIP status icon is yellow (F0010680)
- There is a problem with the DTMF tones. In particular the digit 9 may not be recognised by the
receiving party (F0011378)
- If an iD808 is configured to use a static IP address and this address is changed from i cms,
i cms loses communication with the device until the device is repowered or a configuration
change is made on the device (F0011379)
- If a subcription tries to refresh when the subscription server is not accessible on the
network the subscription fails and does not recover when the server is available again. The
SIP status icon is yellow (F0011492)
- Attempts to use G722 with the Avaya PBX end up with calls using G711 (F0011564)
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New Features Added in Version 1.0.10.0
- The LED associated with a busy elsewhere line is now illuminated red (was previously not illuminated)
Defects Resolved in Version 1.0.10.0
- Occassional kernel panic when upgrading units (F0010629 & F0010719)
- Improve error reporting for network, i cms and SIP server problems. Error messages reported
in "Show Network" and highlighted in red. This includes configuration profile errors, any DSP reset and
i cms out of sync (F0010670)
- When contact is lost with the i cms server an announce is sent every 30 seconds so that
when the i cms server is contactable again the i cms icon is rapidly updated (F0010793)
- With an ARD appearance on a dynamic key and cleared from the far end, while getting the call failed tone clear the
call from the speaker channel (double assign) the ARD appearance icon is initially correct and then changes
to a tick (F0011014)
- A dynamic speaker channel with on ARD or MRD call does not have its icon updated when the call is on a handset
and then cleared from the handset (F0011017)
- After a configuration change the voice recording functionality is only correct after a reset (F0011020)
- No CDR event when switching a call between handsets (F0011021)
- No CDR event(s) received when calls are active and the user logs out (F0011022)
- Dynamic speaker channel icon not updated when a SIP call is moved between handsets (F0011023)
- When using the numbers to navigate the menu, after entering the admin password, the menu for the
selected line is always displayed and not the selected menu (F0011025)
- The SIP server status icon is lit green indicating that an appearance has been registered even
though there are none configured on the unit (F0011038)
- Missing CDR event when assigning a conference to a dynamic speaker channel (F0011039)
- IP header checksum on SbRTP is incorrect once every 65535 packets (F0011040 & F0011163)
- When you try to make a call and no lines are available an error message is given which states "No
line availables" instead of "No line available" (F0011043)
- Cannot take a VPW call off hold (F0011049)
- UI crash when in the outbound dialling mode and the same line is selected again (F0011054 & F0011162)
- A VPW cannot be switched between handsets (F0011055)
- When dialling the Avaya voice mail the key presses on the iD808 have to be slow and deliberate
to avoid errors (F0011056)
- If a call appearance is allocated to the user but not assigned to a key the appearance is not registered
as expected but a subscription attempt is made which fails (F0011058)
- The iD808 lost contact with i cms during an attempt to add a speaker key from i cms (F0011061)
- The upgrader filename should use a capital D in iD808 (F0011067)
- It should be possible to use privacy on one handset with one hidden appearance even when the device
is configured for two handsets (F0011068)
- Errors reported in message log when moving a call between handsets (F0011069)
- Occassionally the page title disappears from the top of screen B (F0011073)
- When adding a new appearance on a page through the iD808, if a page/sub-page is selected that
currently has nothing on it and the user presses the exit button to cancel the addition, the
page displayed in screens B and C will still be the empty page selected when the addition was
to be made. (F0011075)
- The call icon on a dynamic speaker channel is not updated when a connecting call is answered (F0011091)
- Style 9 is almost unreadable as the bright pink background makes it hard to make out what
the white text is (F0011095)
- Sometimes it is not possible to dial out from a VPW because the unit has run out of outgoing lines even
though the VPW was using a call appearance (F0011097)
- The key finder text is misleading when deleting keys types that are always fixed keys e.g. paginating and
speaker keys (F0011100)
- With auto hold enabled you cannot dial after seizing the line for the second call (F0011107)
- Errors are reported if an outgoing call is ringing at the far end and digits are pressed (F0011108)
- When a VPW is edited so that it has a different line ID and then changed back to the original line
ID, the VPW can longer make an outbound calls (F0011127)
- The domain name under Configure Network -> Device IP Addresses displays the address for the NTP server
that the device is using rather than the correct domain name (F0011131)
- ARD calls don’t cleardown cleanly and can remain in the busy elsewhere state (F0011138)
- On the iD808 when in a call initiated from a line appearance you have two way voice. When the
initiator presses the privacy key sometimes there is only one way voice. The voice from the initiator is
no longer present (F0011149)
- Call quality deteriorates over time on G.711 (F0011161)
- VPW calls should not be entered into the call register (F0011169)
- If a SIP call is in the hold state on a dynamic speaker key pressing the dynamic speaker key to
unhold the call leads to an ambiguous state (F0011174)
- Using iD808s set to use G.729 and a Cisco PBX that forces G.722 and with a 3 way
conference on the iD808s and 8 active ARD calls, repeatedly clearing and adding the third iD808 in the
conference can result in one or all of the other iD808s losing all audio (F0011182)
- Making and breaking a point to point sip call in rapid succession can cause a dsp error followed by
dsp restart. Reproduced with G.722 calls (F0011190)
- Pressing transfer on a VPW causes the VPW to be in the connected state but leaves the other end
on hold and the only thing the user can do is clear the call (F0011225)
- During a call transfer attempt on an Asterisk PBX if the call transfer is cancelled the first call
is also cleared. (F0011229)
- Occassional after making, putting on hold and receiving lots of calls on one iD808 putting a point-to-point
call on hold and dialling another call can result in a double ring tone being heard on the handset (F0011233)
- If the iD808 starts up without an i cms server contactable then the screen saver will never
be displayed because the device continues to attempt to send announce messages to i cms (F0011245)
- If the iD808 is set up on a different time zone to GMT then the screen saver can come on as soon as
the iD808 is powered up (F0011246)
- When making a point to point call through the corporate directory and then making the call private on the
calling device, occassionally the audio cuts out on either end of the call so no audio can be heard or
transmitted (F0011255)
- The iD808 uses the subnet mask obtained from the DHCP server in headers sent to i cms even when
using fixed IP (F0011264 & F0011277)
- Occassionally an iD808 gets into a state where it cannot be resynchronised from i cms (F0011268)
- Setup an iD808 with both Cisco and Asterisk call appearances on the Asterisk subnet. Set Cisco to use
G.722 and Asterisk to use G.711A. Set the Cisco appearance as the default. Dial from this iD808 using
the default appearance to another iD808 on the Asterisk subnet. The codec in use should be G.722. Clear
this call and select an Asterisk appearance on the iD808. Dial the other iD808 on the Asterisk subnet.
The codec should be G.711A. After a few seconds audio is lost in both directions and the DSP restarts (F0011273)
- When a device is being synchronised, the user is able to edit the login information and make an attempt
to login despite the edit boxes being greyed out (F0011282)
- The Avaya default call appearance should have priority to be the master call appearance for bridge calls (F0011283)
- The DSP reset counter should not be defaulted when resynchronizing from i cms (F0011284)
- One iD808 with 9 incoming calls from 3 other iD808s, answer one of the calls and you get one way audio
incoming and constant rubbish being played through the receiving handset (F0011298)
- Attempting to put a connecting ARD call on hold puts it in a strange state and causes it to become unusable (F0011311)
- Set up a call between two iD808s. Make a second call to the same iD808 and attempt to answer it on the same
handest as the existing call. "HANDSET BUSY" message box appears. While this message box is active clear
down the active call and now you cannot dial any numbers (F0011320)
- If we have more than 24 SbRTP Channels on appearance keys then any new SIP Call will have no way voice (F0011333)
- Missing CDR events when ARD call is cleared from the far end (F0011352)
- ARD appearance calls on hold drop out. This is because the iD808 stops transmitting (F0011354)
- When handset Push to Talk is enabled and a call is clear the handset mode becomes inconsistent and
shows it to be not muted (F0011363)
- No error or warning message is shown to the user when attempting to assign a fixed speaker channel
(Hoot, ARD or MRD) to a dynamic speaker channel with nothing on (F0011365)
- Incorrect data in CDR events. When enabling/ disabling privacy for ARD/MRD calls the PRIVACY_CHANGE event
message is correctly generated however the privacy request and privacy status flags in the message are
incorrect (F0011366)
- CDR event incorrect when taking an ARD call off hold. The associated audio device is listed as NONE and no voice
recording stream is listed. The correct details are shown when the the call is next moved between handsets
or to a speaker channel (F0011367)
- Error message seen in log file when pressing transfer for a seized line or a connecting call (F0011368)
- Cannot ring an MRD call (the RING soft key is not shown) when in a conference however the soft key is
shown when the MRD call is the only call in the conference or the last call added to the conference (F0011369)
- If the device is out of sync with i cms (i cms icon is yellow) and the user logs out from the device,
the icon stays yellow (F0011372)
- The internal out of sync count is not cleared when the device is logged out. This means that if a device
is re-powered when logged out and i cms is not available the device will start up and attempt
to contact i cms which will fail. When i cms becomes available the icon will only change
to yellow because settings obtained from the DHCP server (such as NTP server address and Domain name)
have not been reported to i cms. When a user logs in, the device becomes in sync and the icon
is corrected (F0011373)
Defects Resolved in Version 1.0.10.0 patch 1
- The DTMF tone for digit 9 may not be recognised by the receiving party (F0011378)
Known Defects/Issues in Version 1.0.10.0
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New Features Added in Version 1.0.9.0
- UI enhanced to give 3D effect on keys
- Updated colours for key and alerting styles
- Updated online help text
- CDRs updated to match latest specification
- ARD on-hold functionality added
- Maximum handset volume increase and the default value changed to 7 (in the UI range 1 to 10)
- Audio device 1 & 2 handset/headset configuration added to user preferences menu and acoustic
settings optimised for a headset when headset is selected.
- Different icons introduced for handset 1 & 2 to allow them to be differentiated and
"speaker channel on handset" icons introduced
Defects Resolved in Version 1.0.9.0
- Bug fix to stop lines already setup for VPWs from being selected when adding speed dials
(F0010698)
- Fix to stop the UI from crashing when selecting a VPW that is configured to use
the appearance that is also the default appearance for that user (F0010625).
- Fixed bug related to dialling a 255 long phone number (F0010676)
- Fixed bug where pressing the privacy button while it is ringing causes two
calls to be mixed together (F0010720)
- Fixed a bug when the user is doing a manual dialling and soft function key is
pressed it changes the call state to line seized state when it should tell
the user that the handset is busy (F0010726)
- Fixed a bug to handle privacy button press when the call is in line seized
state (F0010727)
- Fixed bug that crashed the UI when seating a user from i cms to a device that
already had a user seated (F0010695)
- Fixed network/i cms configuration lost during an upgrade (F0010487)
- Fixed bug where pressing the hash key in the directory list entered the search screen
in numeric mode and not upper case alpha mode (F0010635)
- Fixed a bug where with an active call on a dynamic speaker channel, line seize followed
by assign and then pressing the dynamic speaker channel should drop the seized line and
copy the dynamic speaker channel call to the handset (F0010732 & F0010736)
- On an Avaya or Cisco PBX the caller ID is now updated when the call at the far end is
transferred to another user (F0010525)
- Fixed problem where barging into a call does not record a placed call in the call
register (F0010696)
- Fixed problem when receiving a call and DND is on, if i press DND to be off the unit
does not ring until the initiator hangs up, then the unit rings once and is hung up (F0010721)
- Fixed problem where "Mute alerts now" should immediately silence any playing alerts (F0010730)
- Fixed problem where cannot move ARD between handsets (F0010714)
- Fixed problem while in the call failed state speed dial presses are ignored (F0010728)
- Fixed problem when in the call failed state privacy key presses should be ignored (F0010729)
- Fixed problem where no far end ringing tone is played on the handset when an ARD is
in the connecting state (F0010743)
- With a seized line on both handsets, switch one seized line to the other handset and the seized
line is move but the other seized line is not dropped and can no longer be used (F0010776)
- In outbound dialling pressing message waiting should clear the dial string and dial voice mail
and pressing a speed dial should clear the dial string and dial the speed dial (F0010697)
- Fixed problem while in the connecting state where if a blank dynamic key is pressed a handset
busy warning is generated (F0010733)
- Fixed problem where a call that is on a dynamic speaker channel can be put on hold but is not
shown as being on hold (F0010735)
- Error conditions are shown in red on the "Show Network" screen (F0010670)
- Fixed problem where if you try to delete an entry from the personal directory that is linked
to a speed dial on a non-paginating key on the iD808 the unit no longer recognises any other
key presses and stays stuck in "are you sure you want to delete" (F0010770)
- Improvements to the updating of the i cms status icon. The iD808 now continues to send an
announce every 30 seconds if the last announce failed and will update the status icon when an
announce is successful (F0010793)
- When a handset has an Avaya seized line on it and is selected, selecting another call attempts
to drop the seized line to reuse the handset but the seized line is not cleared and eventually
times out at the PBX and is then stuck in the call failed state (F0010792)
- Fixed inconsistent use of upper case in the warning dialogs e.g. "on Other Handset" should
be "on other handset" (F0010820)
- With a message waiting, receive an incoming call on the default appearance but do not answer
the call. Clear the call from the far end. Press the message waiting key and listen to the
"Welcome to Audix" message. Press a digit and the user is taken to outbound dialling instead
of playing the DTMF tone to line (F0010850)
- When using redial on an iD808, if the handset has single line information the line that has been
seized is shown rather than the number dialled (F0010780)
- Changed the corners of the warning dialog from white to black (F0010794)
- If an appearance is configured to not allow outbound calls any attempt to manual dial, seize
a line, speed dial, dial a VPW or dial from a directory using that line should be rejected
with an "Outbound calls restricted" warning dialog (F0010791)
- Seize Avaya call appearance then select the menu and then press the same call appearance
key clears the seized line (F0010738)
- Adding the first call to a conference (putting a conference on hold) does not update the soft
keys (F0010863)
- The UI crashes when receiving a call from a Cisco PBX (F0010690)
- Disabled manual and speed dialling of one digit numbers (F0010692)
- Manual dialling not validated correctly when editing dial string (F0010693)
- In a 3 way conference can get one way or no-way voice (F0010581 & F0010757)
- First time asterisk call is put on hold the music on hold keeps on cutting out. Take the call
off hold and then put it back on hold it works fine (F0010689)
- Fixed DSP crash when using G.729 (F0010737)
- When an ARD call is on a handset the far end ring tone can be heard on all local desk
stations (iD808s or iD114s) (F0010772)
- When dialing a SIP call the "far end ringing" tone is heard on the handset. If the call is
moved to a dynamic speaker channel then the ringing tone is not moved and error messages
are generated (F0010778)
- Key press tone enhanced (F0010786)
- When the iD808 cannot contact the i cms server it now sends out an announce every
30 seconds until successful so that the i cms status indicator is returned to
green within 30 seconds of the connection being restored (F0010793)
- Having a SIP call on a dynamic speaker channel then adding it to a conference with a Hoot call
on a normal speaker channel causes the icon next to the dynamic speaker channel not to get
updated when the conference is put on hold (F0010795)
- With a SIP call and a Hoot call (on a speaker channel) in a conference, the conference cannot
be added to a dynamic speaker channel. This should be allowed in all cases (F0010796)
- Flash can get filled up after many upgrades (old zips are not deleted during an upgrade) (F0010804)
- With a connected call on handset 1, a failed call on handset 2, handset 1 selected and auto
hold enabled, pressing the failed call line key correctly puts the connected call on hold but
the failed call is moved to the handset when it should of been cleared and a new line seize
attempt should be initiated on the handset (F0010815)
- With a connected call on handset 1, outbound dialling on handset 2, handset 1 selected
(therefore outbound dialling is hidden) and auto hold enabled, pressing the outbound dialling
line key correctly puts the connected call on hold but the outbound dialling is not moved
to handset 1 (F0010816)
- With a connected call on the selected handset the redial button should always bring up the
redial menu (unless there are no numbers to redial). Attempts to dial from the redial menu
need to check the auto hold mode and depending on the mode display handset busy or put the
connected call on hold. (F0010817)
- Pressing the redial button when there are no numbers to redial should warning the user that
there are no numbers to redial. (F0010818)
- When talking on the second leg of a transfer, assign the call to a speaker channel and then
press the clear button for the handset that was previously being used. This acts as if the
call was still attached to the handset. (F0010819)
- "Transfer incomplete, original call disconnected" popup is required to warn the user that
the first leg of the call has been cleared. This can happen if the far end clears or another
use picks up the on hold call. Also if the transfer outbound dialling menu is being displayed
this should be automatically removed. (F0010821)
- Transfer button press should be rejected for barged in calls and the user should be warned
"Transfer denied for a barged in call" (F0010822)
- Hold button press should be rejected during transfer operation and the user should be warned
"Hold denied during transfer" (F0010823)
- Pressing privacy in a conference should generate a not allowed popup (F0010827)
- With auto hold disabled, an active call on the selected handset and an incoming bridged call
ringing on a dynamic key, press the dynamic key and handset busy should be displayed but it
is not (F0010830)
- Need to block assigning to speaker when the call on the handset is the line seize for the
second leg of a transfer operation (F0010832)
- Need to block any line key or message waiting key presses when the call on the handset is
the line seize for the second leg of a transfer operation. Handset busy should be displayed (F0010833)
- Seize a line and press the line key for a second time clears the line seize (F0010834)
- If you dial an invalid number for the second leg of a transfer the call fails immediately and
the first leg is automatically reinstated to the handset. The required behaviour is to keep
the first leg on hold and warning the user of the failed call attempt with the normal call
failed state on the handset. (F0010835)
- If during the second leg of a call transfer the far end disconnects the call the handset
should go to the call failed state and the first leg should not be automatically reinstated
to the handset. The only situation where the first leg is automatically reinstated is when
the local user clears the second leg of the call. (F0010836)
- When a speaker channel call is on a handset the audio device icon shown against the speaker
channel should be a "speaker-on-handset" icon (F0010870)
- A hoot channel without any talk rights must be inhibited from joining a conference with a
popup warning (F0010871)
- When a call that is on-hold and attached to a dynamic speaker channel is cleared from the
far end the speaker channel should go to the call failed state before being cleared (F0010872)
- When a SbRTP channel is moved to a handset configured in single line display mode the handset
shows the reference label for the channel instead of the short label. (F0010873)
- Create a conference with an ARD call (speaker or appearance) as the only call in the conference.
The call cannot be selected when it is in the conference on hold state (F0010876)
- With a call on hold and on a dynamic speaker channel, assign+assign+the dynamic speaker channel
key should clear the on hold call off the dynamic speaker channel but leave the call on hold
on the appearance key (F0010883)
- With a call on hold and on a dynamic speaker channel, assign+the dynamic speaker channel key
should take the call off hold and copy the call to the handset (F0010884)
- With G.722 and G.729 after waiting some time in a point-to-point call the voice is distorted (F0010915 & F0010937)
- When in a full conference the initiator can not put the conference on hold but can put it on
a speaker channel. If you remove one user from the conference and then the initiator presses
hold the conference goes onto a speaker channel and now cannot be assigned back to the handset. (F0010917)
- Pressing a key for a VPW that worked in V1.0.8.2 now states "Outbound calls disabled" in V1.0.8.25 (F0010919)
- Incorrect audio setup for Hoot channel in a conference on a speaker channel. With a Hoot
channel on hold selecting the channel opens the microphone and other deskstations can hear
what is said but nothing can be heard from the other deskstations to the iD808. Assigning
the conference to the handset allows two way voice again. (F0010932)
- Cannot answer an incoming call on a VPW key when using the Avaya PBX (F0010941)
- Paginating soft function keys lose their text when a paginating call appearance rings (F0010956)
- Point to point SIP call on one handset and an ARD on the other handest. Cancel the ARD from
the far end and a call failed tone is heard on the handset connected to the SIP call (F0010959)
- When initiating an ARD straight to a handset no voice is heard from the initiating phone until
the handset button is pressed even though it is set as push to mute (F0010961)
- When selecting an empty dynamic speaker channel in the idle state, the caution box "Outbound
calls restricted" is displayed. It should do nothing (F0010962)
- Cannot assign a call onto a dynamic speaker channel that contains a failed call. The correct
operation should be to clear the failed call and move the call (F0010965)
- Microphone indicator not updated (always on) when initiating an ARD call on a speaker channel
and while it is still connecting, assign it to the handset. The microphone indicator is correct
when assigning a connected ARD call to the handset (F0010966)
- There is a compatibility problem between iD808 and i-series devices where 1 packet in every 65536
is being rejected by the i-series device (recorded as a "Bad IP header packet error count") (F0011007)
- When Push to Talk is enabled on the handset and the call is moved to speaker free, the mute should be disabled (F0011012)
- After upgrading an iD808 the ethernet switch IC in the iD808 is operating in half duplex mode until
the unit is repowered and this can cause packet loss that can affect voice quality and can cause ARD
calls to drop out (F0011009)
- If voice mail option is selected in main menu and a busy line is selected (e.g. speed dial, active call,
hoot etc.) (voice mail option disappears if not selected) the selected line should change as the voice
mail option is not available (F0011013)
- When using adding a speed dial or virtual private wires the outbound id option should not list the avaya
hidden appearances (F0011015)
- Info menu item not grayed out when MRD/ARD appearance also on dynamic speaker cleared from handset while
looking at the main menu. (pressing handset clear again does correct it) (F0011016)
- No ARD off hold action. When an ARD call is put on hold and then taken off hold and put back on the handset
and another user joins the call the call changes to busy elsewhere and the user has to exit the call and
re-enter to talk again. (F0011019)
Known Defects/Issues in Version 1.0.9.0
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New Features Added in Version 1.0.8.0
- Common lamping with an Avaya PBX now fully implemented
- Improved audio acoustics and audio sub-system can also be adjusted using CM
variables for audio optimisation experiments
- Two environment configurations supported (Trader and Office) which can be selected
from the "User Preferences" menu
- Implemented a minimum MRD ring time of 5 secs (F0010158)
- Rich text formatting added to online help
- Handset sidetone support added
- Red status icons now flash
- Updated online help text
- DND softkey text modified
- Updated icons and styles
Defects Resolved in Version 1.0.8.0
- Fixed problem where call event (missed, placed, received call) sometimes not
entered into call register and the error "Invalid subscription index" is logged
to the messages file (F0010582)
- Removed incorrect error message "Invalid call state" generated by
UI_far_end_ringing_from_mtfif when making an external call (F0010583)
- Fixed a problem where hanging up on each phone individually rather than clearing
the conference from the iD808. When the last person exits the conference, their
appearance on the iD808 locks up with a red cross against their number (F0010578)
- Fix bug that resulted in incorrect information being stored in the call register
(F0010572)
- Fixed a bug related to audio streams disappear or go to the wrong destination
when multiple SIP calls are moved around on a device, via handsets and Speaker
Channels (F0010618)
- Changed behaviour of DND. When operating with an Avaya PBX DND simply silences
incoming alerts. When operating with a non-Avaya PBX DND will forward an incoming
call if call forward on busy is enabled and if not enabled will reject the call
(F0010543)
- Changes to support parameters of h1 and h2 to identify the hidden call appearances
for handset 1 and 2 respectively (F0010453)
- Fixed excessive audible clicks heard with lost packets (F0010213)
- Fixed Ethernet problem where DSP error messages occurs causing no way voice (F0010531)
- Fixed problem where the second G.729 call has poor voice quality (F0010577)
- Fixed a UI crash when creating a VPW (F0010625)
- Now validates the MAC address for all profile downloads from i cms. This
is to stop the profile from being delivered to the wrong device if that
device obtains an IP address that used to belong to a different device (F0010202)
- Show "Caller ID Withheld" for incoming call with no caller ID and stop users
from dialling these calls from the call register (F0010430)
- The absence of a default appearance is now only consider an error that is shown
as a yellow i cms icon if there is at least one appearance configured for
the unit (F0010489)
- Fixed problem where no caller ID shown when operating with a Cisco PBX (F0010536)
- Fixed problem that did not show handset busy warning when attempting to pick up
a call on a dynamic key (F0010537)
- Line ID list in the key edit screen incorrectly formatted if the reference label
for ringdown circuit was left blank in i cms (F0010538)
- Fixed error in operation when moving ARD calls between handsets (F0010539)
- Domain name not always shown correctly in the "Show Network" screen (F0010541)
- When editing a profile the preview ringtone can be left playing (F0010545)
- Incorrect labels shown for an ARD call on a dynamic key (F0010549)
- Linux host name is now set correctly (F0010556)
- An incoming call to an Avaya call appearance can be presented to two dynamic
keys when paging (F0010557)
- Attempts to send CDR events can create increasing numbers of sockets until no more
sockets can be created at which point the UI cannot render any icons (F0010561)
- Fixed problem that caused the UI to lock up when selecting handsfree or
conferencing (F0010532)
- Fixed problem where the last subscription index entry for an appearance was
not initialised and as a result Avaya FNU calls such as call forwarding
interfere with other keys for that appearance (F0010520)
- Fixed problem where the displayed caller ID is different for incoming calls
received via notified messages and those received via invite messages (F0010519)
- Fixed problem where an incoming call that is received with both an invite
and a notify message is presented to two dynamic keys when the call appearance
is on a hidden page (F0010526)
- Use of hidden appearances with the Avaya PBX changed so that the initial call
(prior to privacy being applied) uses the public call appearance (F0010395 &
F0010415)
- Styling change to the online help
- Fix bug that stopped CM variables being restored in the upgrader script
- Added CDR heartbeat
- Handset sidetone can now be configured correctly
- G722 operation in telephony calls fixed
- UI crash when using call transfer on an Avaya PBX fixed (F0010413)
- Call failed tone fixed when an ARD call is on hands-free
- Soft keys now correctly updated when an ARD call is cleared from the far end
- Fixed problem where common lamping calls are not recorded in the correct call register file
(F0010434)
- Fixed problems related to call transfer and support added to cancel call transfer when either
second call is ringing or established (F0010431)
- Fixed problem related to change of remote MAC Address on a voice recording link
- Fixed problem to now reject a privacy request for a non-Avaya PBX
- Fixed problem where a NTP server IP address change from i cms was not seen until after
a power cycle (F0010155)
- Fix problem when moving a bridged call from a handset to a speaker channel when the
associated line appearance is on a paginating key
- Fixed problem with bridged call operation where the text shown on some keys is incorrect
- Correct the mis-spelling of the word "configuration" in the CM application (F0010265)
- Fix a problem that stopped music-on-hold being heard when operating with a Cisco PBX
- Fix a problem that sometimes resulted in the start of the welcome message not being heard
when contacting a voice mail server (F0009426)
- Fix a problem where incoming calls on common lamping appearances played an alerting tone
even when the appearance key had alerting turned off (F0010336)
- Fix a problem where an ARD call that is configured as non-latching can be left latched open
(F0010191 & F0010200)
- ARP requests for remote MAC addresses changed to solve problem when proxy ARP is not
available. The local gateway MAC address is used as remote MAC address
- Now correctly displays the remote identity on the line and handset keys when barging into a call
- Now correctly displays "Conference" as the remote identity for a conference call
- Fixed problem that resulted in slow UI response when the unit was trying to contact a DNS
server that is not available on the network
- Fixed problem that a key was deleted if moved to itself
- Fix problem where two or more synchronises from iManager in close succession can result
in the directories not being loaded (F0010272)
- Fixed problem where subscription registration is lost after one or two days
- Fixed problem where no text is displayed when returning to the confirmation screen from
the online help for directory address delete and directory address copy
Known Defects/Issues in Version 1.0.8.0
- CDR logging only partially tested
- Noise reduction not implemented
- Styling (colours) are expected to change
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New Features Added in Version 1.0.7.0
- Updated screen saver
- Modified operation for locating i cms server using DNS.
Now uses SRV server requests for _speakerbus-cms-1._tcp and
_speakerbus-cms-2._tcp to retrieve the FQDN of the primary
and backup servers
Defects Resolved in Version 1.0.7.0
- Fixed problem with hold and transfer when operating with a Cisco PBX
- Modified displayed text for call/line appearances, VPW, handset keys,
dynamic keys and dynamic speaker channel keys to meet the recently
agreed requirements
- Fix UI crash when using VPW keys
- Fix problem where iD808 freezes when editing a line style/profile on the
default call appearance after hitting save (F0010157)
- Fix potential problem of losing a configuration in i cms when there is a
communication problem between i cms and the iD808. The iD808 no longer sends
corrections back to i cms but will now log the error and change the i
cms server status icon to yellow. (F0010043)
- Fix to support asymmetric call settings on SIP calls (F0009468)
- Fix problem when sending DTMF tones to line. The microphone is now muted
while the tone is being sent so that the microphone audio is not mixed
with the tone (F0010159)
- Fix problem that stopped i cms changes to the i cms server
settings from taking immediate affect. Also sends an announce if the
settings are changed from the device.
- Fix problem that reset the DSP when using multiple SbRTP channels
Known Defects/Issues in Version 1.0.7.0
- CDR logging only partially tested
- Common lamping only partially implemented
- Noise reduction not implemented
- Handset sidetone not implemented
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New Features Added in Version 1.0.6.0
- Support added for line seize when operating with an Avaya PBX
Defects Resolved in Version 1.0.6.0
- Fixed problem where the iD808 was using the previous stored DNS address when the
iD808 was configured with a static IP address and no DNS server address (F0009909, F0009910)
- Fixed possible UI crash when unregistering lines during logout
- Fixed problem with handling of i cms close socket protocol (F0009641, F0009642)
- Fixed problem with i cms out of sync messaging
- Corrected problem with volume for key press tones
Known Defects/Issues in Version 1.0.6.0
- CDR logging only partially tested
- Problems with hold when operating with a Cisco PBX
- Problems with transfer when operating with a Cisco PBX
- Common lamping only partially implemented
- Noise reduction not implemented
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New Features Added in Version 1.0.5.0
- Partial implementation of common lamping when operating with an Avaya PBX. Features
supported include call status icon and indicator update for a bridged call appearance,
answering of a ringing elsewhere line, barge-in to a busy elsewhere line, making an
outgoing call on a bridged call appearance and taking a held elsewhere call off-hold.
Features not yet implemented include line seize, updating text for a bridged line,
updating the call register and displaying a ringing elsewhere line on a dynamic key.
Privacy is only partially implemented
- Handsfree speaker volume is limited to avoid howl when handsfree is selected at both
ends of a point-to-point call
- Status icons added for missed calls, call forward active and do not disturb active
- Enhanced upgrader support to retain the network and i cms configuration during
an upgrade. (Note this only becomes active when upgrading from this version or later)
- Handset volume adjustment added to user preferences menu
- Screen saver added
- Support for LogOffSync from i cms added
- Support for interworking with i series 3.0
Defects Resolved in Version 1.0.5.0
- Fixed call forwarding with an Avaya PBX
- Fixed problem that stopped call appearances being displayed in the outbound ID list
when programming a speed dial
- Fixed problem when logging in from i cms when the user is in an edit box in the
login screen (F0009652)
- Fixed problem with handset audio quality
Known Defects/Issues in Version 1.0.5.0
- CDR logging only partially tested
- Problems with hold when operating with a Cisco PBX
- Problems with transfer when operating with a Cisco PBX
- Common lamping only partially implemented
- Noise reduction not implemented
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New Features Added in Version 1.0.4.0
- When operating with Avaya PBXs only call appearances are registered
Defects Resolved in Version 1.0.4.0
- Fix problem where the keys are not updated correctly when the remote end clears a
call that is on a speaker channel if the appearance key for the call is located on
a key that could be programmed as a speaker key
- Fix problem that there is no way to retrieve a call once transfer is press if the
user decides not to transfer the call (F0009581)
- Fix problem where the user could dial another call when a 2-way conference on the
selected handset. This erroneous call would fail and the user cannot cancel this
call or the conference (F0009594)
- Fix problem that caused the SIP stack to crash or calls to fail when transferring or
holding call on a PBX using authentication
- Fix problem where text on soft function keys dissappears when entering add line key
finder mode and when changing pages in add line key finder mode
- Fix problem that resulted in button presses on speaker channel 1 being ignored (F0009586)
- Fix problem with handling audio packet sizes other than 20ms (F0009468)
- Fix operation of received voice activity indication on speaker channels (F0009591)
- Fix problem where the title of a page can be edited by the user even when the page is
set to read-only from i cms (F0009627)
- Fix problem in the DSP jitter buffer code that could result in a crash of the DSP (F0009587)
Known Defects/Issues in Version 1.0.4.0
- CDR logging only partially tested
- Problems with hold when operating with a Cisco PBX
- Problems with transfer when operating with a Cisco PBX
- Call forwarding when operating with an Avaya PBX not supported yet
- Common lamping when operating with an Avaya PBX not supported yet
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New Features Added in Version 1.0.3.0
- Added feature to provide music-on-hold (if supported by PBX) during conference
hold if there is only single SIP call in the conference call (F0009419)
- Diagnostic facility added to enable DSP debug commands and messages on the serial interface
Defects Resolved in Version 1.0.3.0
- No password protection on the RS232 terminal interface
- Latching mode on dynamic speaker channels not correct for all call types
- Handset style not replaced when ARD call cleared during a conference call
- Fix error resulting in the default appearance not being correctly configured
by i cms
- Fix bug that can result in a blank time and date being displayed instead of
"Clock error" when the NTP server cannot be contacted
- Fix error where the latching mode may be incorrect when moving telephony
calls to a dynamic speaker channel
- Fix error where keys are not repainted with their non-key-finder colours
when leaving key-finder after moving a line key (F0009429)
- Fix error where gooseneck mic active indicator is shown in the wrong state
- Fix crash on registration with incorrect authentication password
- When updating keys from the iD808 menu system all the parameters associated
with the change including those that haven't changed value are sent to
i cms otherwise i cms can reject the change
- Echo canceller was inverted and therefore incorrectly disabled by default
- Fix problem where the rotary volume knobs were controlling the wrong speaker channel
- Fix problem where handset key is not returned to its original style when moving
outbound dialling between handsets
- Fix problem where cursor is not shown when moving outbound dialling between handsets
- Fix problem where call transfer is not possible when there is only a single (call or line) appearance
- Fix problem where call transfer, music on hold is not heard at the far end until
the number to transfer to has been entered (F0009418)
- Fix following problem. Select a line to dial out on, start dialling but do not complete
dial string. Then select the other handset, continue dialling. The iD808 will ignore the
first numeric key pressed after the second handset is selected (F0009424)
- Fix problem where pressing the conference key whilst viewing a profile disables the exit key (F0009428)
- Fix problem when transferring a call to a different call appearance on the same iD808 (F0009421 & F0009422)
- Fix problem when pressing the conference key to put an outgoing call on the conference, the manual dialing
does not work
- Echo canceller performance improved (optimised for the gooseneck microphone, but still requires
optimisation for handsfree operation)
- Fix DSP problem so that the DSPs can handle 10 G.729 Tx and Rx streams and 8 SbRTP Tx and
128 SbRTP Rx streams
- Fix a problem that can cause an occasional lock up in the DSP
- Corrections to colours used for alerting style 8
Known Defects/Issues in Version 1.0.3.0
- CDR logging only partially tested
- Problems with hold when operating with a Cisco PBX
- Problems with transfer when operating with a Cisco PBX
- Problems with call forwarding when operating with an Avaya PBX
- Problems with transfer when operating with an Avaya PBX
- Echo canceller requires optimisation to improve performance in handsfree mode
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Defects Resolved in Version 1.0.2.0
- "Locate i cms via DNS" operation fixed
- Fixed out of sequence registration error sometimes seen with an Avaya PBX
- User is warned with a splash screen if the unit is reset from i cms
- Line seized followed by selecting the voice mail menu option operation corrected
- Correction to operation of message waiting indicator
- Fixed error when unregistering lines from Avaya PBX
- Fixed error when when using a filename that includes a path when doing a firmware
download from i cms
- Fixed error in the voice recording that resulted in the hands-free channel voice being
recorded on both the handset 1 and the handset 2 recording streams
- Fix for ARD/MRD calls to use the new alert style when a different profile is selected
for a key
- Fix for ARD/MRD calls to remove them from a dynamic key when cleared from the far end
- Correction to gooseneck microphone level indicators to correctly interpret the data
from the DSP
- Check made, when selecting a dynamic key, that the key has something on it before
displaying the handset busy message or putting the call on hold
- Calls that are still ringing are now added to a free dynamic key when the SIP or ARD/MRD
call on the dynamic key is answered or cleared
- Soft function key text is now updated when exiting from key finder mode
- Ringing indicator added for ARD/MRD calls
- Priority check made before determining which of the still ringing keys should be displayed
on the dynamic key. The order is priority then duration of ring for calls with equal
priority
- Echo canceller functionality included
- Fixed bug in Transmitted RTP packet where the sequence number would be incorrectly
incremented when an RTCP packet was sent (causing a lost packet report)
- Tidied up DSPLink startup to reliably start both DSP processors
- Dynamic keys updated when a call is added to a conference
- In a select Line ID menu, blank lines are now highlighted and not ignored when selected
- Unit now works correctly after a "Device IP Address" change for the system menu
- Fixed voice recording channel that contain a mix of calls using different audio codecs
- Fixed UI lockup when the unit is logged out by i cms while in the admin
login menu
- Correctly handle an empty profile upload request(to upload the entire profile)
Known Defects/Issues in Version 1.0.2.0
- CDR logging only partially tested
- Problems with hold when operating with a Cisco PBX
- Problems with transfer when operating with a Cisco PBX
- Problems with call forwarding when operating with an Avaya PBX
- Problems with transfer when operating with an Avaya PBX
- Latching mode on dynamic speaker channels not correct for all call types
- Handset style not replaced when ARD call cleared during a conference call
- No password protection on the RS232 terminal interface
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This is the first release of iD808 and contains the following
features:
iD808 General
- iD808 main unit only (no support for expansion units in this release)
- Physically the unit has two Ethernet ports but for this release only
one of these ports is utilised in normal operation
- The unit has two USB ports but for this release neither of the ports
are utilised in normal operation
- Supports one or two handsets with a built-in volume control and a push
button that can be configured for push-to-talk or push-to-mute
- Supports an internal microphone (used for hands-free operation)
- Supports a gooseneck microphone (used for talking on speaker channels)
- Supports an internal speaker with a master volume control that is used
for hands-free operation, speaker channels (they have associated volume
knobs) and audible alerts
- Configuration via i cms
- Configuration via a build-in menu system
iD808 User Interface
- English only for this release
- Online help incorporated into the menu system
- Dial-pad for manual dialling and use in the menu system
- Assign key (for moving calls between handsets and speaker channels)
- Privacy key for preventing barge-in and conferencing. Only supported
by ringdown calls
- Redial key for telephony calls
- Conference key for conferencing hoot, ringdown and telephony calls in any
combination. Supports up to a 6-way conference
- Transfer key for transferring telephony calls
- Hold keys for handsets 1 and 2 (to put calls on-hold)
- Clear keys for handsets 1 and 2 (to clear down calls unless they are
attached to a speaker channel in which case they are returned to the
speaker channel)
- Navigation keys (up,down,left,right,ok & back) to navigate within the
menu system and to move between pages and sub-pages when not in the menu
system
- Page key to move to the next sub-page
- Speaker key to cancel handsfree operation or to put the selected handset
into handsfree mode
- Support for 100 key pages. Each key page can contain up to 64 key entries
although there is a maximum limit of 600 entries spread over the 100 pages.
Each page has a customizable title
- There are two phonebooks directories for telephony calls; a read-only corporate
directory and a writeable personal directory. The corporate directory can
contain up to 5000 entries whereas the personal directory can contain up to
600 entries. Each entry can hold up to 10 telephone numbers. Entries and
individual address can be copied from the corporate directory to the
personal directory. Entries and addresses in the personal directory can be
edited, copied or deleted or new entries or addresses can be created
- The menu system supports add, edit, view, move and delete of all key types
except that handset keys and dynamic keys cannot be added or deleted. This
uses key finder to locate keys. Speed dial and VPW keys can also be added from
the phonebook directories using either key finder or freekey wizard. Keys can
be protected from modification from i cms
- The menu system supports editing alert profiles, setting the default alert
profile and configuring the alerts overrides. The menu system also supports
changing the alert settings for keys or pages of keys
- The menu system supports configuring user settings and call settings
- Time and date synchronised to an NTP server
- The user is able to log in or log out of a unit. Log in requires a user ID
and password. There is a separate password for administrator access to the
network configuration and engineering tools screens
iD808 Soft Key Types
- Paginating key. These keys can take on the identity of any other soft key types
depending on which page/sub-page is being shown with the exception of handset
keys and the dynamic keys which are not allowed to be paginating
- Line appearance key. An external telephony line
- Call appearance key. An internal telephony extension for the logged in user
- Bridged call appearance key. An internal telephony extension for another user
(not the logged in user)
- Anonymous call appearance key. For incoming calls only; used to display an
incoming call that cannot be matched to any of the VPW, line, call or bridged
call appearances. These calls would otherwise be rejected if no anonymous call
appearances existed on the unit
- Speed dial key (for telephony calls)
- Virtual private wire (VPW) key. These are a simulation of a dedicated
person-to-person line using SIP. This combines a dedicated line and a speed
dial onto a single key
- Automatic ringdown (ARD) appearance key. An i series ARD line appearance
- Manual ringdown (MRD) appearance key. An i series MRD line appearance
- Hoot appearance key. An i series hoot line appearance
- Dynamic key. To show a "ringing" appearance from a hidden page for telephony,
ARD or MRD calls
- Menu shortcuts key. A shortcut to the main directory menu or to the main
call register menu
- Page shortcuts key. A shortcut to specific key pages
- Handset key. A key for handset 1 or handset 2
- Soft key. A context sensitive key. Can display any of the following functions:
MRD ring, temporary mute of alerts, voicemail message waiting, call forward
menu & do not disturb
iD808 Speaker Channel Key Types
- Dynamic Speaker Channel. Telephony, hoot, ARD, MRD and conference calls may
be moved to these channels using the assign key
- Static Speaker Channel. Hoot, ARD and MRD channels whose home is on a specific
speaker channel key. They can be temporarily moved to a handset but the speaker
channel is never freed up for any other use
iD808 Call Types
- Telephony. Uses the SIP protocol. Always via a PBX (has been tested with Cisco,
Avaya and Asterisk based PBXs). SIP calls will not be possible without a PBX present.
Coding types supported are G711-Alaw, G711ulaw, G729, and G722. The requested coding
mode is configurable on a per unit basis
- Hoot, ARD and MRD. i series SbRTP protocol. Supports flexible SbRTP with packet
sizes of 1,2 & 4ms and VAD on or off. Also supports local and global muting and privacy.
Requires i series v3.0 to support privacy (to stop additional users barging into or
conferencing into a point-to-point call) and to provide support for flexibility SbRTP
- The maximum number of active calls is 10 of any mix e.g. 10 SIP calls or 8 hoot channels +
2 ARD channels or 7 SIP calls + 1 hoot + 1 ARD + 1 MRD etc.) The calls will reside on
the 2 handsets and the 8 speaker channels. One of the 2 handsets can be in hands-free mode.
In hands-free mode the handset is inactive and the internal microphone and speaker are used
along with a built-in echo canceller. One of the 10 calls may be a conference call that
supports a maximum of a six-way conference call by mixing the audio locally. Hence with
1 six-way conference call and 9 other calls active there will be a total of 14 active
call legs
- In addition to the number of active calls there may also be a large number of "ringing"
calls including ARD and MRD calls. The total number of hoot plus ARD plus MRD channels that
are registered via IGMP on a single unit is 60 but the i cms configuration may need
to further limit this number depending on the SbRTP configuration and the number of gateways
in use etc. to ensure that the 100MB Ethernet link is not overloaded. The maximum number of
"ringing" SIP calls is 600
Telephony Features
- Manual dialling using a default appearance
- Line seize followed by manual dialling
- Speed-dial using specified outbound ID
- Line seize followed by speed dial
- Dial from directory using default appearance
- VPW dial using its dedicated line appearance
- Call register logging
- Re-dial
- Dial from call register
- Line appearance matching for incoming calls
- Caller ID directory entry matching for incoming calls
- Conferencing (PBX independent - the iD808 is also able to conference SIP calls to
ARD/MRD/hoot channels. Only one conference can be supported at any one time and the
maximum size of this conference is a six-way conference (5 far end locations plus this unit).
The conference is mixed locally on the iD808 and does not use the SIP conference protocol
- Voice mail support
- Transfer (Attended & Unattended)
- Divert / Call Forwarding (No-Answer, On-Busy, All). This will only divert / call forward
calls that are delivered on a call appearance (i.e. not calls delivered on line appearances or
bridged call appearances)
- Hold
- Call waiting
- DTMF Support (in-band only)
- Message Waiting Indication (MWI)
- Auto-Hold (independent of the IP-PBX). Automatically put any active call on the selected
handset on hold when the user answers or makes a new call
- Auto-Swap (independent of the IP-PBX). Automatically put any active call on the selected
handset on hold when the user takes another call off hold
- Auto-Handset-Mute (independent of the IP-PBX). Automatically mutes an active call on a
handset when making or answering a call on the other handset
Alerting
- Each appearance or static speaker channel (except for hoot channels) can be configured
to have alerting active or inactive with a specified alert profile number
- There are 32 alert profiles that can be chosen. Each alert profile determines the ring-tone,
the display style, the mode, volume and priority for playing "ring" for incoming telephony
calls or ringing ARD and MRD channels. The priority determines which ring-tone is played
when there are multiple calls/channels that are simultaneously generating ring
- Alerts can be temporarily muted for the duration of a call
- Alerts can be overridden to be either all off or all on
- Alerting can be configured to ring on busy
- Alerting can be configured to override the alerting state to on for the currently displayed page
- Alerts that are overridden to on use a default alert profile. The default alert profile can
be configured by the user
- Dynamic keys show alerting keys that are hidden (not on the currently displayed page / sub-page)
sorted in order of alert priority followed by the duration of the alert. All the dynamic keys
are refreshed when a user changes page / sub-page otherwise a dynamic key showing an alerting
key remains static until that key stops alerting. When alert ringtones are played the ringtone
played is the ringtone of the highest priority alert even if that alert is attached to a key on
a hidden page which can happen if all dynamic keys are occupied
Voice Recording
- Three audio IP streams with audio mixed from a combination of handsets and speaker channels
can be sent to a voice recorder as G.711/G.729 RTP streams
- CDR call logging. All call events are sent to a call logger via a TCP stream using a
proprietary protocol
Miscellaneous Features
- Supports Autodiscovery
- Supports SNMP (MIB-II only)
- Supports IGMP for registering multicast channels but does not act as a querier
- Ping available from menu system
- "Safe Mode" available from menu system. When a unit cannot contact i cms a safe mode
menu can be accessed by entering "safemode" as the User ID. This allows the network and i cms
server configurations to be checked and modified so that i cms can be successfully
contacted
- Key press tones can be enabled to provide the user with an audible feedback for each key press
Known Defects/Issues in Version 1.0.1.0
- No echo canceller functionality
- i cms interworking not tested
- CDR logging only partially tested
- Telephony does not operate correctly and the UI can crash
after changing the "Device IP Address" configuration and
logging out the unit
- Problems with hold when operating with a Cisco PBX
- Problems with transfer when operating with a Cisco PBX
- Problems with call forwarding when operating with an Avaya PBX
- Problems with transfer when operating with an Avaya PBX
- Updating of dynamic keys sometimes incorrect
- Latching mode on dynamic speaker channels not correct for all call types
- Handset style not replaced when ARD call cleared during a conference call
- For the voice recording streams the hands-free channel voice is recorded on both
the handset 1 and the handset 2 recording streams
- A voice recording channel that should contain a mix of calls using different audio
codecs does not include all the calls
- Voice Recording does not reinitialize correctly after a DSP reset
- No password protection on the RS232 terminal interface
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